Administrator’s Guide for the
SoundPoint® IP/SoundStation® IP
Family
SIP 2.2.2
November, 2007 Edition
1725-11530-220 Rev. A1
SIP 2.2.2
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About This Guide
The Administrator’s Guide for the SoundPoint IP / SoundStation IP family is
for administrators who need to configure, customize, manage, and
troubleshoot SoundPoint IP / SoundStation IP phone systems. This guide
covers the SoundPoint IP 301, 320, 330, 430, 501, 550, 600, 601, and 650 desktop
phones, and the SoundStation IP 4000 conference phone.
The following related documents for SoundPoint IP / SoundStation IP family
are available:
•
•
Quick Start Guides, which describe how to assemble the phones
Quick User Guides, which describe the most basic features available on
the phones
•
•
User Guides, which describe the basic and advanced features available on
the phones
Developer’s Guide, which assists in the development of applications that
run on the SoundPoint IP / SoundStation IP phone’s Microbrowser
•
•
Technical Bulletins, which describe workarounds to existing issues
Release Notes, which describe the new and changed features and fixed
problems in the latest version of the software
For support or service, please contact your Polycom® reseller or go to Polycom
Polycom recommends that you record the phone model numbers, software
(both the bootROM and SIP), and partner platform for future reference.
SoundPoint IP / SoundStation IP models: ___________________________
BootROM version: ________________________________________________
SIP Application version: ___________________________________________
Partner Platform: _________________________________________________
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Contents
BootROM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–3
Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–4
Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–5
Resource Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–7
Available Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–8
Setting Up the Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–2
Setting Up the Boot Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–12
Upgrading SIP Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–17
Setting Up Basic Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–1
Call Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–3
Call Timer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–3
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Administrator’s Guide SoundPoint IP / SoundStation IP
Called Party Identification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–4
Missed Call Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–4
Do Not Disturb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–8
Local Digit Map . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–12
Speed Dial . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–13
Time and Date Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–13
Idle Display Animation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–14
Call Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–16
Call Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–17
Call Forward . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–18
Directed Call Pick-Up . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–19
Group Call Pick-Up . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–20
Call Park/Retrieve . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–20
Last Call Return . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–20
Setting Up Advanced Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–20
Multilingual User Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–27
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Contents
Downloadable Fonts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–28
Microbrowser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–28
Voice Mail Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–30
Multiple Registrations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–31
Server Redundancy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–34
Presence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–37
Voice Activity Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–43
Audio Codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–45
Comfort Noise Fill . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–46
IP Type-of-Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–46
IEEE 802.1p/Q . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–47
Setting Up Security Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–47
Custom Certificates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–48
Status Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–4
Log Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–4
Reading a Boot Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–7
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Administrator’s Guide SoundPoint IP / SoundStation IP
Access to Screens and Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–11
Calling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–12
Master Configuration Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–2
Application Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–4
Protocol <volpProt/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–6
Dial Plan <dialplan/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–16
Localization <lcl/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–19
Tones <tones/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–24
Sound Effects <se/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–28
Voice Settings <voice/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–34
Web Server <httpd/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–54
Directory <dir/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–58
Presence <pres/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–60
Fonts <font/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–60
Keys <key/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–63
Bitmaps <bitmap/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–65
Indicators <ind/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–65
Security <sec/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–73
License <license/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–74
Request <request/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–76
Feature <feature/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–77
Resource <res/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–78
USB Port <usb/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–83
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Contents
Per-Phone Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–82
Registration <reg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–83
Calls <call/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–87
Diversion <divert/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–90
Dial Plan <dialplan/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–92
Flash Parameter Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–99
RFC and Internet Draft Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–2
Request Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–3
Header Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–4
Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–9
Third Party Call Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–9
Trusted Certificate Authority List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–1
Encrypting Configuration Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–3
Adding a Background Logo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–5
Multiple Key Combinations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–9
Default Feature Key Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–10
Parsing Vendor ID Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–16
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Administrator’s Guide SoundPoint IP / SoundStation IP
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1
Introducing the SoundPoint IP /
SoundStation IP Family
This chapter introduces the SoundPoint IP / SoundStation IP family, which is
supported by the software described in this guide.
The SoundPoint IP / SoundStation IP family provides a powerful, yet flexible
IP communications solution for Ethernet TCP/IP networks, delivering
excellent voice quality. The high-resolution graphic display supplies content
for call information, multiple languages, directory access, and system status.
The SoundPoint IP / SoundStation IP family supports advanced functionality,
including multiple call and flexible line appearances, HTTPS secure
provisioning, presence, custom ring tones, and local conferencing.
The SoundPoint IP / SoundStation IP phones are end points in the overall
network topology designed to interoperate with other compatible equipment
including application servers, media servers, internet-working gateways,
voice bridges, and other end points
The following models are described:
•
— IP 301
— IP 320/330
— IP 430
— IP 501
— IP 550
— IP 600/601
— IP 650
•
— IP 4000
This chapter also lists the key features available on the SoundPoint IP /
SoundStation IP phones running the latest software.
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Administrator’s Guide SoundPoint IP / SoundStation IP
SoundPoint IP Desktop Phones
This section describes the current SoundPoint IP desktop phones. For
individual guides, refer to the product literature available at
http://www.polycom.com/support/voice/. Additional options are also
available. For more information, contact your Polycom distributor.
The currently supported desktop phones are:
•
SoundPoint IP 301
•
SoundPoint IP 320/330
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Introducing the SoundPoint IP / SoundStation IP Family
•
SoundPoint IP 430
•
SoundPoint IP 501
•
SoundPoint IP 550
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Administrator’s Guide SoundPoint IP / SoundStation IP
•
SoundPoint IP 600/601
•
SoundPoint IP 650
SoundStation IP Conference Phone
This section describes the current SoundPoint IP conference phone. For
individual guides, refer to the product literature available at
http://www.polycom.com/support/voice/. Additional options are also
available. For more information, contact your Polycom distributor.
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Introducing the SoundPoint IP / SoundStation IP Family
The currently supported conference phone is:
SoundStation IP 4000
•
Key Features of Your SoundPoint IP / SoundStation IP
Phones
The key features of the SoundPoint IP / SoundStation IP phones are:
•
Award winning sound quality and full-duplex speakerphone or
conference phone
— Permits natural, high-quality, two-way conversations (one-way,
monitor speaker in the SoundPoint IP 301)
— Uses Polycom’s industry leading Acoustic Clarity Technology
•
Easy-to-use
— An easy transition from traditional PBX systems into the world of IP
— Up to 18 dedicated hard keys for access to commonly used features
— Up to four context-sensitive soft keys for further menu-driven
activities
•
•
Platform independent
— Supports multiple protocols and platforms enabling standardization
on one phone for multiple locations, systems and vendors
— Polycom’s support of the leading protocols and industry partners
makes it a future-proof choice
Field upgradeable
— Upgrade SoundPoint IP / SoundStation IP as standards develop and
protocols evolve
— Extends the life of the phone to protect your investment
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Administrator’s Guide SoundPoint IP / SoundStation IP
— Application flexibility for call management and new telephony
applications
•
Large LCD
— Easy-to-use, easily readable and intuitive interface
— Support of rich application content, including multiple call
appearances, presence and instant messaging, and XML services
— 4 line x 20 character monochrome LCD for the SoundPoint IP 301
— 102 x 23 pixel graphical LCD for the SoundPoint IP 320/330
— 160 x 80 pixel graphical grayscale LCD for the SoundPoint IP 501
— 320 x 160 pixel graphical grayscale LCD for the SoundPoint IP
550/600/601/650 (supports Asian characters)
— 248 x 68 pixel graphical LCD for the SoundStation IP 4000
•
•
Dual auto-sensing 10/100baseT Ethernet ports
— Leverages existing infrastructure investment
— No re-wiring with existing CAT 5 cabling
— Simplifies installation
Power over Ethernet (PoE) port
— Unused pairs on Ethernet port pairs are used to deliver power to the
phone via a wall adapter allowing fewer wires to desktop
— Optional accessory cable for CiscoR Inline Powering and IEEE 802.3af
on the SoundPoint IP 301 and SoundPoint IP 501
— Built-in PoE on the SoundPoint IP 550, 600, 601, and 650 (auto-sensing)
•
Multiple language support
— Set on-screen language to your preference. Select from Chinese,
Danish, Dutch, English, French, German, Italian, Japanese, Korean,
Norwegian, Portuguese, Russian, Spanish, and Swedish
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2
Overview
This chapter provides an overview of the Session Initiation Protocol (SIP)
application and how the phones fit into the network configuration.
SIP is the Internet Engineering Task Force (IETF) standard for multimedia
conferencing over IP. It is an ASCII-based, application-layer control protocol
(defined in RFC 3261) that can be used to establish, maintain, and terminate
calls between two or more endpoints. Like other voice over IP (VoIP)
protocols, SIP is designed to address the functions of signaling and session
management within a packet telephony network. Signaling allows call
information to be carried across network boundaries. Session management
provides the ability to control the attributes of an end-to-end call.
For the SoundPoint IP / SoundStation IP phones to successfully operate as a
SIP endpoint in your network, it must meet the following requirements:
•
•
•
•
A working IP network is established.
Routers are configured for VoIP.
VoIP gateways are configured for SIP.
The latest (or compatible) SoundPoint IP / SoundStation IP phone SIP
application image is available.
•
A call server is active and configured to receive and send SIP messages.
http://www.polycom.com/techpartners1/ .
This chapter contains information on:
•
•
•
To install your SoundPoint IP / SoundStation IP phones on the network, refer
SoundStation IP phones with the desired features, refer to Configuring Your
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Administrator’s Guide SoundPoint IP / SoundStation IP
SoundStation IP phones on the network, refer to Troubleshooting Your
Where SoundPoint IP / SoundStation IP Phones Fit
The phones connect physically to a standard office twisted-pair (IEEE 802.3)
10/100 megabytes per second Ethernet LAN and send and receive all data
using the same packet-based technology. Since the phone is a data terminal,
digitized audio being just another type of data from its perspective, the phone
is capable of vastly more than traditional business phones. AsSoundPoint IP /
SoundStation IP phones run the same protocols as your office personal
computer, many innovative applications can be developed without resorting
to specialized technology.
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Overview
Session Initiation Protocol Application Architecture
The software architecture of SIP application is made of 4 basic components:
•
•
•
•
BootROM—loads first when the phone is powered on
Application—software that makes the device a phone
Configuration—configuration parameters stored in separate files
Resource Files—optional, needed by some of the advanced features
Configuration
Resource
Files
bootROM
Application
BootROM
The bootROM is a small application that resides in the flash memory on the
phone. All phones come from the factory with a bootROM pre-loaded.
The bootROM performs the following tasks in order:
1. Performs a power on self test (POST).
2. (Optional) Allows you to enter the setup menu where various network on
provisioning options can be set.
The bootROM software controls the user interface when the setup menu is
accessed.
3. Requests IP settings and accesses the boot server to look for any updates
to the bootROM application.
If updates are found, they are downloaded and saves to flash memory,
eventually overwriting itself after verifying the integrity of the download.
4. If a new bootROM is downloaded, format the file system clearing out any
application software or configuration files that may have been present.
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Administrator’s Guide SoundPoint IP / SoundStation IP
5. Download the master configuration file.
This file is either called <mac-address>.cfg or 000000000000.cfg . This file
is used by the both the bootROM and the application for a list of other files
that are needed for the operation of the phone.
6. Examine the master configuration file for the name of the application file,
and then look for this file on the boot server.
If the copy on the boot server is different than the one stored in flash
memory or, if there is no file stored in flash memory, the application file is
downloaded.
If the Application is any SIP version prior to 1.5, the bootROM will also download all
the configuration files that are listed in the master configuration file.
Note
7. Extract the application from flash memory.
8. Install the application into RAM, then upload a log file with events from
the boot cycle.
The bootROM will then terminate, and the application takes over.
Application
The application manages the VoIP stack, the digital signal processor (DSP), the
user interface, and the network interaction. The application managed
everything to do with the phone’s operation.
The application is a single file binary image and, as of SIP 1.5, contains a digital
signature to prevent tampering or loading or rogue software images.
If your phones are using bootROM 3.0 or later, the application must be signed.
Warning
All SIP 1.5 applications and later are signed, but later patched versions of 1.3 and
1.4 support this feature. Refer to the latest Release Notes to verify if the image is
signed.
There is a new image file in each release of software.
The application performs the following tasks in order:
1. Downloads system and per-phone configuration files and resource files.
These files are called sip.cfg and phone1.cfg by default. You can
customized the filenames.
If the Application is any SIP version prior to 1.5, the bootROM would have
downloaded all the configuration files that are listed in the master configuration file.
Note
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Overview
2. Controls all aspects of the phone after it has restarted.
3. Uploads log files.
BootROM and Application Wrapper
Both the bootROM and the application run on multiple platforms (meaning all
previously released versions of hardware that are still supported).
The file stored on the boot server is a wrapper, with multiple hardware specific
images contained within. When a new bootROM or application is being saved,
the file is read until a header matching the hardware model and revision are
found, and then only this image is saved to flash memory.
Configuration
The SoundPoint IP / SoundStation IP phones can be configured automatically
through files stored on a central boot server, manually through the phone’s
local UI or web interface, or a combination of the automatic and manual
methods.
The recommended method for configuring phones is automatically through a
central boot server, but if one is not available, the manual method will allow
changes to most of the key settings.
The phone configuration files consist of:
•
•
Configuration files should only be modified by a knowledgeable system
Warning
administrator. Applying incorrect parameters may render the phone unusable. The
configuration files which accompany a specific release of the SIP software must be
used together with that software. Failure to do this may render the phone unusable.
Master Configuration Files
The master configuration files can be one of:
•
•
•
Specified master configuration file
Per-phone master configuration file
Default master configuration file
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Administrator’s Guide SoundPoint IP / SoundStation IP
Application Configuration Files
Typically, the files are arranged in the following manner although parameters
may be moved around within the files and the filenames themselves can be
changed as needed. These files dictate the behavior of the phone once it is
running the executable specified in the master configuration file.
The application files are:
•
Application—It contains parameters that affect the basic operation of the
phone such as voice codecs, gains, and tones and the IP address of an
application server. All phones in an installation usually share this category
of files. Polycom recommends that you create another file with your
organization’s modifications. If you must change any Polycom templates,
back them up first. By default, sip.cfg is included.
•
Per-phone—It contains parameters unique to a particular phone user.
Typical parameters include:
— display name
— unique addresses
Each phone in an installation usually has its own customized version of
user files derived from Polycom templates. By default, phone1.cfg is
included.
Central Provisioning
The phones can be centrally provisioned from a boot server through a system
of global and per-phone configuration files. The boot server also facilitates
automated application upgrades, logging, and a measure of fault tolerance.
Multiple redundant boot servers can be configured to improve reliability.
In the central provisioning method, there are two major classifications of
configuration files:
•
•
System configuration files
Per-phone configuration files
Parameters can be stored in the files in any order and can be placed in any
number of files. The default is to have 2 files, one for per-phone setting and one
for system settings. The per-phone file is typically loaded first, and could
contain system level parameters, letting you override that parameter for a
given user. For example, it might be desirable to set the default CODEC for a
remote user differently than for all the users who reside in the head office. By
adding the CODEC settings to a particular user’s per-phone file, the values in
the system file are ignored.
Verify the order of the configuration files. Parameters in the configuration file loaded
first will overwrite those in later configuration files.
Note
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Overview
The following figure shows one possible layout of the central provisioning
method.
Manual Configuration
When the manual configuration method is employed, any changes made are
stored in a configuration override file. This file is stored on the phone, but a
copy will also be uploaded to the central boot server if one is being used. When
the phone boots, this file is loaded by the application after any centrally
provisioned files have been read, and its settings will override those in the
centrally provisioned files.
This can create a lot of confusion about where parameters are being set, and so
it is best to avoid using the manual method unless you have good reason to do
so.
Resource Files
In addition to the application and the configuration files, the phones may
require resource files that are used by some of the advanced features. These
files are optional, but if the particular feature is being employed, these files are
required.
Some examples of resource files include:
•
•
Language dictionaries
Custom fonts
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•
•
•
Ring tones
Synthesized tones
Contact directories
Available Features
Any new features introduced after SIP 2.1.2 are not supported on the
SoundPoint IP 300 and 500.
Note
This section provides information the features available on the SoundPoint IP
/ SoundStation IP phones:
•
Basic Features
automatic off-hook call placement feature for each .
calls to another destination.
phone for another task, such as making or receiving another call.
identification, time and date, and call duration in three separate lists,
missed calls, received calls, and placed calls on most platforms.
be retrieved by any phone.
maintained for each distinct call in progress.
progress to some other destination.
on another call, the incoming call is presented to the user visually on
the display and a configurable sound effect will be mixed with the
active call audio.
of the party specified for outgoing calls.
derived from the network signalling, when an incoming call is
presented, if information is provided by the call server.
user has connected is displayed and logged, if the name is provided
by the call server.
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Overview
sound effects, such as the ringer, and the receive volume of call audio
is adjustable.
incoming call alerting and other indications are customizable.
phone can be picked up by dialing the extension of the other phone.
Calls to another phone within a pre-defined group can be picked up
without dialing the extension of the other phone.
types.
apply distinctive treatment to calls containing specific attributes.
and the ring type for specific callers can be assigned in the contact
directory.
stop all incoming call alerting.
standard with a handset and a dedicated headset connection (not
supplied). The SoundPoint IP 320, 330, 430, 500, 501, 550, 600, 601, and
650 and SoundStation IP 4000 phone are full-duplex speakerphones.
The SoundPoint IP 301 phone is a listen-only speakerphone.
display a customized animation on the idle display in addition to the
time and date.
together the local user with the remote parties of two independent
calls and can support centralized conferences for which external
resources are used such as a conference bridge.
directory that can be downloaded from the boot server and edited
locally.
setup phase of number-only calls.
indicator (MWI) LED when instant messages and voice messages are
waiting.
visual feedback is provided.
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missed since the user last looked at the Missed Calls list.
extensive use of intuitive, context-sensitive soft key menus.
from dedicated keys as well as from a speed dial menu.
operating modes such as when the phone is idle and during a call.
•
Advanced Features
unavailable and allows ACD login and logout. Requires call server
support.
logically related to each other. Requires call server support.
party information of users through the busy lamp field (BLF) LEDs
and displays on an attendant console phone. Requires call server
support.
from the factory defaults.
be customized by changing the fonts and graphic icons used on the
display and the LED indicator patterns.
messages.
phones and the SoundStation IP 4000 phone support an XHTML
microbrowser.
Integration—SoundPoint IP and SoundStation IP phones can used
with Microsoft Live Communications Server 2005 and Microsoft
Office Communicator to help improve business efficiency and
increase productivity and to share ideas and information immediately
with business contacts. Requires call server support.
have multilingual user interfaces.
calls. The hold feature can be used to pause activity on one call and
switch to another call.
allocated to a single .
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Overview
phone. (SoundStation IP 4000 supports a single .)
types of network address translation (NAT).
users/devices and allows other users to monitor it. Requires call
server support.
transport protocol (RTP) streams as bi-directional from a control
perspective and expects that both RTP end points will negotiate the
respective destination IP addresses and ports.
deployments to ensure continuity of phone service for events where
the call server needs to be taken offline for maintenance, the server
fails, or the connection from the phone to the server fails.
logically related to each other. Requires call server support.
and efficient audible call progress feedback generated by the PSTN
and traditional PBX equipment, call progress tones are synthesized
during the life cycle of a call. Customizable for certain regions, for
example, Europe has different tones from North America.
•
Audio Features
cancellation for hands-free operation.
the transmit gain of the local user in certain circumstances.
operation, reduces background noise to enhance communication in
noisy environments.
to the remote user of a hands-free call.
a standard RTP-compatible technique for conveying DTMF dialing
and other telephony events over an RTP media stream.
(DTMF) tones in response to user dialing on the dial pad.
with an 802.1Q VLAN header.
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high-performance jitter buffer and packet error concealment system
designed to mitigate packet inter-arrival jitter and out-of-order or lost
(lost or excessively delayed by the network) packets.
latency for audio packet transmission.
detecting periods of relative “silence” in the transmit data path and
replacing that silence efficiently with special packets that indicate
silence is occurring.
•
Security Features
menus are protected with two privilege levels, user and
administrator, each with its own password.
configuration files must be protected (encrypted). The phone can
recognize encrypted files, which it downloads from the boot server
and it can encrypt files before uploading them to the boot server.
server for application provisioning, the phone trusts certificates
issued by widely recognized certificate authorities (CAs).
validating incoming network signaling.
For more information on each feature and its associated configuration
4-1.
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3
Setting up Your System
Your SoundPoint IP / SoundStation IP SIP phone is designed to be used like a
regular phone on a public switched telephone network (PSTN).
This chapter provides basic instructions for setting up your SoundPoint IP /
SoundStation IP phones. This chapter contains information on:
•
•
•
•
Because of the large number of optional installations and configurations that
are available, this chapter focuses on one particular way that the SIP
application and the required external systems might initially be installed and
configured in your network.
For more information on configuring your system, refer to Configuring Your
For installation and maintenance of Polycom SoundPoint IP phones, the use of a
boot server is strongly recommended. This allows for flexibility in installing,
upgrading, maintaining, and configuring the phone. Configuration, log, and directory
files are normally located on this server. Allowing the phone write access to the
server is encouraged.
Note
The phone is designed such that, if it cannot locate a boot server when it boots up,
it will operate with internally saved parameters. This is useful for occasions when
the boot server is not available, but is not intended to be used for long-term
operation of the phones.
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Setting Up the Network
Regardless of whether or not you will be installing a centrally provisioned
system, you must perform basic TCP/IP network setup, such as IP address
and subnet mask configuration, to get your organization’s phones up and
running.
The bootROM application uses the network to query the boot server for
upgrades, which is an optional process that will happen automatically when
properly deployed. For more information on the basic network settings, refer
The bootROM on the phone performs the provisioning functions of
downloading the bootROM, the <Ethernet address>.cfg file, and the SIP
Basic network settings can be changed during bootROM download using the
bootROM’s setup menu. A similar menu system is present in the application
for changing the same network parameters. For more information, refer to
DHCP or Manual TCP/IP Setup
Basic network settings can be derived from DHCP, or entered manually using
the phone’s LCD-based user interface, or downloaded from configuration
files.
Polycom recommends using DHCP where possible to eliminate repetitive manual
data entry.
The following table shows the manually entered networking parameters that
may be overridden by parameters obtained from a DHCP server, an alternate
DHCP server, or configuration file:
Alternate
DHCP
Configuration File
(application only)
Local
FLASH
DHCP Option
Parameter
DHCP
D priority when more than one source exists D
1
2
3
4
IP address
subnet mask
IP gateway
1
1
3
•
•
•
-
-
-
-
-
-
•
•
•
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Setting up Your System
Alternate
DHCP
Configuration File
(application only)
Local
FLASH
DHCP Option
Parameter
DHCP
Refer to DHCP
Menu on page
•
•
-
•
boot server
address
151
•
•
-
-
•
Note: This value
is configurable.
SIP server address
SNTP server
address
42 then 4
-
•
•
SNTP GMT offset
2
6
•
•
-
-
•
-
•
•
DNS server IP
address
alternate DNS
server IP address
6
•
•
-
-
-
-
•
•
DNS domain
15
Refer to DHCP
Menu on page
Warning: Cisco Discovery Protocol (CDP) overrides Local FLASH
that overrides DHCP VLAN Discovery.
VLAN ID
http://www.ietf.org/rfc/rfc2131.txt?number=2131 or
http://www.ietf.org/rfc/rfc2132.txt?number=2132.
The configuration file value for SNTP server address and SNTP GMT offset can
be configured to override the DHCP value. Refer to
A-51.
Note
The CDP value can be obtained from a connected Ethernet switch if the switch
supports CDP.
In the case where you do not have control of your DHCP server or do not have
the ability to set the DHCP options, an alternate method of automatically
secondary DHCP server that responds to DHCP INFORM queries with a
requested boot server value is one possibility. For more information, refer to
http://www.ietf.org/rfc/rfc3361.txt?number=3361 and
http://www.ietf.org/rfc/rfc3925.txt?number=3925.
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Supported Provisioning Protocols
The bootROM performs the provisioning functions of downloading
configuration files, uploading and downloading the configuration override
file and user directory, and downloading the dictionary and uploading log
files.
The protocol that will be used to transfer files from the boot server depends on
several factors including the phone model and whether the bootROM or SIP
application stage of provisioning is in progress. By default, the phones are
shipped with FTP enabled as the provisioning protocol. If an unsupported
protocol is specified, this may result in a defined behavior (see the table below
for details of which protocol the phone will use). The Specified Protocol listed
in the table can be selected in the Server Type field or the Server Address can
include a transfer protocol, for example http://usr:pwd@server (refer to
domain string name, or URL. The boot server address can also be obtained
through DHCP. Configuration file names in the <Ethernet address>.cfg file
can include a transfer protocol, for example
https://usr:pwd@server/dir/file.cfg. If a user name and password are
specified as part of the server address or file name, they will be used only if the
server supports them.
A URL should contain forward slashes instead of back slashes and should not
contain spaces. Escape characters are not supported. If a user name and
password are not specified, the Server User and Server Password will be used
Note
Protocol used by
bootROM
Protocol used by
SIP Application
301, 320, 330, 430,
501, 550, 600, 601,
650, 4000
301, 320, 330, 430,
501, 550, 600, 601,
650, 4000
Specified
Protocol
FTP
FTP
FTP
TFTP
HTTP
HTTPS
TFTP
HTTP
HTTP
TFTP
HTTP
HTTPS
There are two types of FTP methods—active and passive. As of SIP 1.5 (and
bootROM 3.0), the SIP application is no longer compatible with active FTP. At that
time, secure provisioning was implemented.
Note
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Setting up Your System
Setting Option 66 to tftp://192.168.9.10 has the effect of forcing a TFTP download.
Using a TFTP URL (for example, tftp://provserver.polycom.com) has the same
effect.
Note
For downloading the bootROM and application images to the phone, the
secure HTTPS protocol is not available. To guarantee software integrity, the
bootROM will only download cryptographically signed bootROM or
application images. For HTTPS, widely recognized certificate authorities are
trusted by the phone and custom certificates can be added (refer to Trusted
Modifying the Network Configuration
You can access the network configuration menu:
•
During bootROM Phase. 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.
•
During Application Phase. The network configuration menu is accessible
from the phone’s main menu. Select Menu>Settings>Advanced>Admin
Settings>Network Configuration. Advanced Settings are locked by
default. Enter the administrator password to unlock. The factory default
password is 456.
Phone network configuration parameters may be modified by means of:
•
•
•
•
•
Use the soft keys, the arrow keys, the Select and Delete keys to make changes.
Certain parameters are read-only due to the value of other parameters. For
example, if the DHCP Client parameter is enabled, the Phone IP Addr and
Subnet Mask parameters are dimmed or not visible since these are guaranteed
to be supplied by the DHCP server (mandatory DHCP parameters) and the
statically assigned IP address and subnet mask will never be used in this
configuration.
Resetting to Factory Defaults
The basic network configuration referred to in the following sections can be
reset to factory defaults using a multiple key combination described in
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Main Menu
The following configuration parameters can be modified on the main setup
menu:
Name
Possible Values
Description
DHCP Client
Enabled, Disabled
If enabled, DHCP will be used to obtain the parameters
3-2.
DHCP Menu
Note: Disabled when DHCP client is disabled.
Phone IP Address
Subnet Mask
dotted-decimal IP address
Phone’s IP address.
Note: Disabled when DHCP client is enabled.
dotted-decimal subnet
mask
Phone’s subnet mask.
Note: Disabled when DHCP client is enabled.
IP Gateway
dotted-decimal IP address
Phone’s default router.
Server Menu
SNTP Address
dotted-decimal IP address
OR
Simple Network Time Protocol (SNTP) server from
which the phone will obtain the current time.
domain name string
GMT Offset
-13 through +12
Offset of the local time zone from Greenwich Mean
Time (GMT) in half hour increments.
DNS Server
dotted-decimal IP address
dotted-decimal IP address
domain name string
Primary server to which the phone directs Domain
Name System (DNS) queries.
DNS Alternate Server
Secondary server to which the phone directs Domain
Name System queries.
DNS Domain
Ethernet
Phone’s DNS domain.
EM Power
Enabled, Disabled
This parameter is relevant if the phone gets Power over
Ethernet (PoE). If enabled, the phone will set power
requirements in CDP to 12W so that up to three
Expansion Modules (EM) can be powered. If disabled,
the phone will set power requirements in CDP to 5W
which means no Expansion Modules can be powered (it
will not work).
Syslog
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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.
Note
Note
The EM Power parameter is only available on SoundPoint IP 601 and 650 phones.
To switch the text entry mode on the SoundPoint IP 330/320, press the #. You may
want to use URL or IP address modes when entering server addresses.
DHCP Menu
The DHCP menu is accessible only when the DHCP client is enabled. The
following DHCP configuration parameters can be modified on the DHCP
menu:
Possible
Name
Values
Description
Timeout
1 through 600
Number of seconds the phone waits for secondary DHCP Offer
messages before selecting an offer.
Boot Server
0=Option 66
The phone will look for option number 66 (string type) in the
response received from the DHCP server. The DHCP server
should send address information in option 66 that matches one
of the formats described for Server Address in the following
phone sends out a DHCP INFORM query and the following
scenarios are possible:
•
If no alternate DHCP server responds:
- The INFORM query process will retry and eventually time
out.
- The boot server value stored in flash will be used.
•
A single alternate DHCP server responds. This is
functionally equivalent to the scenario where the primary
DHCP server responds with a valid boot server value.
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Possible
Name
Values
Description
Boot Server (continued)
1=Custom
The phone will look for the option number specified by the Boot
Server Option parameter (below), and the type specified by
the Boot Server Option Type parameter (below) in the
response received from the DHCP server. If the DHCP server
sends nothing, the phone sends out a DHCP INFORM query
and the following scenarios are possible:
•
If no alternate DHCP server responds:
- The INFORM query process will retry and eventually time
out.
- The boot server value stored in flash will be used.
•
A single alternate DHCP server responds. This is
functionally equivalent to the scenario where the primary
DHCP server responds with a valid boot server value.
2=Static
The phone will use the boot server configured through the
Server Menu. For more information, refer to the following
3=Custom+Option
66
The phone will first use the custom option if present or use
Option 66 if the custom option is not present. If the DHCP
server sends nothing, the phone sends out a DHCP INFORM
query and the following scenarios are possible:
•
•
If no alternate DHCP server responds:
- The INFORM query process will retry and eventually time
out.
- The boot server value stored in flash will be used.
A single alternate DHCP server responds.
- The phone prefers the custom option value over the
Option 66 value, but if no custom option is given, the phone
will use the Option 66 value. This is functionally equivalent
to the scenario where the primary DHCP server responds
with a valid boot server value.
Boot Server Option
128 through 254
(Cannot be the
same as VLAN ID
Option)
When the boot server parameter is set to Custom, this
parameter specifies the DHCP option number in which the
phone will look for its boot server.
Boot Server Option Type
0=IP Address,
1=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. The IP Address must specify
the boot server. The String must match one of the formats
Menu.
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Possible
Values
Name
Description
VLAN Discovery
0=Disabled
(default)
No VLAN discovery through DHCP.
1=Fixed
Use predefined DHCP vendor-specific option values of 128,
144, 157 and 191. If this is used, the VLAN ID Option field will
be ignored
2=Custom
Use the number specified in the VLAN ID Option field as the
DHCP private option value.
VLAN ID Option
128 through 254
(Cannot be the
same as Boot
Server Option)
The DHCP private option value (when VLAN Discovery is set
to Custom).
For more information, refer to Assigning a VLAN ID Using
(default is 129)
If multiple alternate DHCP servers respond:
Note
•
•
The phone should gather the responses from alternate DHCP servers.
If configured for Custom+Option66, the phone will select the first response that
contains a valid "custom" option value.
•
If none of the responses contain a "custom" option value, the phone will select
the first response that contains a valid “option66” value.
Server Menu
The following server configuration parameters can be modified on the Server
menu:
Name
Possible Values
Description
Server Type
0=FTP, 1=TFTP, 2=HTTP,
3=HTTPS, 4=FTPS, 5=Invalid
The protocol that the phone will use to obtain
configuration and phone application files from the boot
Note: Active FTP is not supported for bootROM version
3.0 or later. Passive FTP is still supported.
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Name
Possible Values
Description
Server Address
dotted-decimal IP address
OR
domain name string
OR
The boot server to use if the DHCP client is disabled, the
DHCP server does not send a boot server option, or the
Boot Server parameter is set to Static. The phone can
contact multiple IP addresses per DNS name. These
redundant boot servers must all use the same protocol. If
a URL is used it can include a user name and password.
directory and the master configuration file can be
specified.
URL
All addresses can be followed
by an optional directory and
optional file name.
Note: ":", "@", or "/" can be used in the user name or
password these characters if they are correctly escaped
using the method specified in RFC 1738.
Server User
any string
any string
The user name used when the phone logs into the server
(if required) for the selected Server Type.
Note: If the Server Address is a URL with a user name,
this will be ignored.
Server Password
The password used when the phone logs in to the server
if required for the selected Server Type.
Note: If the Server Address is a URL with user name and
password, this will be ignored.
File Transmit Tries
Retry Wait
1 to 10
Default 3
The number of attempts to transfer a file. (An attempt is
defined as trying to download the file from all IP
addresses that map to a particular domain name.)
0 to 300
Default 1
The minimum amount of time that must elapse before
retrying a file transfer, in seconds. The time is measured
from the start of a transfer attempt which is defined as the
set of upload/download transactions made with the IP
addresses that map to a given boot server's DNS host
name. If the set of transactions in an attempt is equal to or
greater than the Retry Wait value, then there will be no
further delay before the next attempt is started.
Provisioning
Method
Default or SAS-VP
If SAS-VP is selected, provisioning is done (in addition to
the normal process).
Network
Cable/DSL,
LAN,
Dial-up
The network environment the phone is operating in.
The default value is Cable/DSL.
Tag SN to UA
Disabled, Enabled
If enabled, the phone’s serial number (MAC address) is
included in the User-Agent header of the Microbrowser.
The default value is Disabled.
Provisioning String
any string
The URL used in XML post/response transactions. If
empty, the configured URL is used.
This field is disabled when Provisioning Method is
Default.
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The Server User and Server Password parameters should be changed from the
default values. Note that for insecure protocols the user chosen should have very
few privileges on the server.
Note
Ethernet Menu
The following Ethernet configuration parameters can be modified on the
Ethernet menu:
Name
Possible Values
Description
CDP
Enabled, Disabled
If enabled, the phone will use CDP. It also reports PoE
power usage to the switch. The default value is Enabled.
VLAN ID
Null, 0 through 4094
Phone’s 802.1Q VLAN identifier. The default value is Null.
Note: Null = no VLAN tagging
VLAN Filtering
Enabled, Disabled
Filter received Ethernet packets so that the TCP/IP stack
does not process bad data or too much data.
Enable/disable the VLAN filtering state.
The default value is Enabled.
Storm Filtering
LAN Port Mode
Enabled, Disabled
Filter received Ethernet packets so that the TCP/IP stack
does not process bad data or too much data.
Enable/disable the DoS storm prevention state.
The default value is Enabled.
0 = Auto
The network speed over the Ethernet.
The default value is Auto.
1 = 10HD
2 = 10FD
3 = 100HD
4 = 100FD
PC Port Mode
Auto, 10HD, 10FD, 100HD,
100FD
The network speed over the Ethernet.
The default value is Auto.
The LAN Port Mode and PC Port Mode parameters are only available on
SoundPoint IP 330, 430, 550, 601, and 650 phones. HD means half duplex and FD
means full duplex.
Note
It is recommended that you leave the LAN and PC parameters set to Auto.
Syslog Menu
Syslog is a standard for forwarding log messages in an IP network. The term
"syslog" is often used for both the actual syslog protocol, as well as the
application or library sending syslog messages.
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The syslog protocol is a very simplistic protocol: the syslog sender sends a
small textual message (less than 1024 bytes) to the syslog receiver. The receiver
is commonly called "syslogd", "syslog daemon" or "syslog server". Syslog
messages can be sent through UDP, TCP, or TLS. The data is sent in cleartext.
Syslog is supported by a wide variety of devices and receivers. Because of this,
syslog can be used to integrate log data from many different types of systems
into a central repository.
The following syslog configuration parameters can be modified on the Syslog
menu:
Name
Possible Values
Description
Server Address
dotted-decimal IP address
OR
domain name string
The syslog server IP address or host name.
The default value is NULL.
Server Type
None=0,
UDP=1,
TCP=2,
TLS=3
The protocol that the phone will use to write to the syslog
server.
If set to “None”, transmission is turned off, but the server
address is preserved.
Facility
0 to 23
A description of what generated the log message. For
more information, refer to section 4.1.1 of RFC 3164.
The default value is 16, which maps to “local 0”.
Render Level
1 to 6
Specifies the lowest class of event that will be rendered to
syslog. It is based on log.render.leveland can be a
lower value.
Note: Use left and right arrow keys to change values.
Prepend MAC
Address
Enabled, Disabled
If enabled, the phone’s MAC address is prepended to the
log message sent to the syslog server.
Setting Up the Boot Server
The boot server can be on the local LAN or anywhere on the Internet.
Multiple boot servers can be configured by having the boot server DNS name
map to multiple IP addresses. The default number of boot servers is one and
the maximum number is eight. The following protocols are supported for
redundant boot servers: HTTPS, HTTP, and FTP. For more information on the
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All of the boot servers must be reachable by the same protocol and the content
available on them must be identical. The parameters described in section
server will be tried for a file transfer and also how long to wait between each
attempt. The maximum number of servers to be tried is configurable. For more
information, contact your Certified Polycom Reseller.
Be aware of how logs, overrides and directories are uploaded to servers that maps
to multiple IP addresses. The server that these files are uploaded to may change
over time.
Note
If you want to use redundancy for uploads, synchronize the files between servers in
the background.
However, you may want to disable the redundancy for uploads by specifying
specific IP addresses instead of URLs for logs, overrides, and directory in the
MACaddress.cfg .
To set up the boot server:
Use this procedure as a recommendation if this is your first boot server setup.
Note
1. Install boot server application or locate suitable existing server(s).
Polycom recommends that you use RFC-compliant servers.
2. Create account and home directory.
If the provisioning protocol requires an account name and password, the server
account name and password must match those configured in the phones. Defaults
are: provisioning protocol: FTP, name: PlcmSpIp, password: PlcmSpIp.
Note
Each phone may open multiple connections to the server.
The phone will attempt to upload log files, a configuration override file,
and a directory file to the server. This requires that the phone’s account has
delete, write, and read permissions. The phone will still function without
these permissions, but will not be able to upload files.
The files downloaded from the server by the phone should be made
read-only.
Typically all phones are configured with the same server account, but the server
account provides a means of conveniently partitioning the configuration. Give each
account an unique home directory on the server and change the configuration on
an account-by-account basis.
Note
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3. Copy all files from the distribution zip file to the phone home directory.
Maintain the same folder hierarchy.
The distribution zip file contains:
— sip.ld (including a separate one for every supported model)
— sip.cfg
— phone1.cfg
— 000000000000.cfg
— 000000000000-directory~.xml
— SoundPointIP-dictionary.xml
— SoundPointIPWelcome.wav
Refer to the Release Notes for a detailed description of each file in the
distribution.
Boot Server Security Policy
You must decide on a boot server security policy.
Polycom recommends allowing file uploads to the boot server where the security
environment permits. This allows event log files to be uploaded and changes made
by the phone user to the configuration (through the web server and local user
interface) and changes made to the directory to be backed up.
For organizational purposes, configuring a separate log file directory is
recommended, but not required. (For more information on
File permissions should give the minimum access required and the account
used should have no other rights on the server.
The phone's server account needs to be able to add files to which it can write
in the log file directory and the root directory. It must also be able to list files
in all directories mentioned in the [mac].cfg file. All other files that the phone
needs to read, such as the application executable and the standard
configuration files, should be made read-only through file server file
permissions.
Deploying Phones From the Boot Server
You can successfully deploy SoundPoint IP and SoundStation IP phones from
one or more boot servers.
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Setting up Your System
Multiple boot servers can be configured by having the boot server DNS name
map to multiple IP addresses. The default number of boot servers is one and
the maximum number is eight. HTTPS, HTTP, and FTP are supported for
redundant boot servers.
To deploy phones from the boot server:
For more information on encrypting configuration files, refer to Encrypting
Note
Note
1. (Optional) Create per-phone configuration files by performing the
following steps:
This step may be omitted if per-phone configuration is not needed.
a
b
Obtain a list of phone Ethernet addresses (barcoded label on
underside of phone and on the outside of the box).
Create per-phone phone[MACaddress].cfg file by using the
phone1.cfg file from the distribution as templates.
Throughout this guide, the terms Ethernet address and MAC address are used
interchangeable.
Note
c
Edit contents of phone[MACaddress].cfg if desired.
For example, edit the parameters.
2. (Optional) Create new configuration file(s) in the style of sip.cfg by
performing the following steps:
For more information on why to create another configuration file, refer to the
“Configuration File Management on SoundPoint IP Phones” whitepaper at
www.polycom.com/support/voice/ .
Note
For more information, especially on the SIP server address, refer to SIP
Most of the default settings are typically adequate, however, if SNTP
settings are not available through DHCP, the SNTP GMT offset and
(possibly) the SNTP server address will need to be edited for the correct
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local conditions. Changing the default daylight savings parameters will
likely be necessary outside of North American locations.
a
(Optional) Disable the local web (HTTP) server or change its
signalling port if local security policy dictates.
b
Change the default location settings for user interface language and
time and date format.
3. (Optional) Create a master configuration file by performing the following
steps:
a
b
c
Create per-phone or per-platform <Ethernet address>.cfg files by
using the 00000000000.cfg and files from the distribution as templates.
A-2.
Edit the CONFIG_FILES attribute of the <Ethernet address>.cfg files
so that it references the appropriate phone[MACaddress].cfg file.
For example, replace the reference to phone1.cfg with
phone[MACaddress].cfg.
Edit the CONFIG_FILES attribute of the <Ethernet address>.cfg files
so that it references the appropriate sipXXXX.cfg file.
For example, replace the reference to sip.cfg with sip650.cfg.
d
e
Edit the LOG_FILE_DIRECTORY attribute of the <Ethernet
address>.cfg files so that it points to the log file directory.
Edit the CONTACT_DIRECTORY attribute of the <Ethernet
address>.cfg files so that it points to the organization’s contact
directory.
4. Reboot the phones by pressing the reboot multiple key combination.
The bootROM and SIP application modify the APPLICATION
APP_FILE_PATH attribute of the <Ethernet address>.cfg files so that it
references the appropriate sip.ld files.
For example, the reference to sip.ld is changed to 2345-11605-001.sip.ld to
boot the SoundPoint IP 601 image.
At this point , the phone sends a DHCP Discover packet to the DHCP server. This
is found in the Bootstrap Protocol/option "Vendor Class Identifier" section of the
packet and includes the phone’s part number and the bootROM version.
Note
For example, a SoundPoint IP 650 might send the following information:
5EL@
DC?5cSc52*46*(9N7*<u6=pPolycomSoundPointIP-SPIP_6502345-12600-001,1B
R/4.0.0.0155/23-May-07 13:35BR/4.0.0.0155/23-May-07 13:35
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Setting up Your System
5. Monitor the boot server event log and the uploaded event log files (if
permitted).
Ensure that the configuration process completed correctly. All
configuration files used by the boot server are logged.
You can now instruct your users to start making calls.
Upgrading SIP Application
You can upgrade the SIP application that is running on the SoundPoint IP and
SoundStation IP phones in your organization. The exact steps that you
perform are dependent on the version of the SIP application that is currently
running on the phones and the version that want to upgrade to.
The bootROM, application executable, and configuration files can be updated
automatically through the centralized provisioning model. These files are
read-only by default.
Most organization can use the instructions shown in the next section,
However, if your organization has a mixture of SoundPoint IP 300 and/or 500
phones deployed along with other models, you will need to change the phone
configuration files to continue to support the SoundPoint IP 300 and IP 500
phones when software releases SIP 2.2.0 or later are deployed. These models
were discontinued as of May 2006. In this case , refer to Supporting
The SoundPoint IP 300 and 500 phones will be supported on the latest
Warning
maintenance patch release of the SIP 2.1 software stream—currently SIP 2.1.2.
Any critical issues that affect SoundPoint IP 300 and 500 phones will be addressed
by a maintenance patch on this stream until the End of Life date for these products.
Phones should be upgraded to BootROM 4.0.0 for these changes to be effective.
Supporting SoundPoint IP and SoundStation IP Phones
To automatically update:
1. Back up old application and configuration files.
The old configuration can be easily restored by reverting to the backup
files.
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2. Customize new configuration files or apply new or changed parameters
to the old configuration files.
Differences between old and new versions of configuration files are
explained in the Release Notes that accompany the software. Both
mandatory and optional changes may present. Changes to site-wide
configuration files such as sip.cfg can be done manually, but a scripting
tool is useful to change per-phone configuration files.
The configuration files listed in CONFIG_FILES attribute of the master configuration
file must be updated when the software is updated. Any new configuration files
must be added to the CONFIG_FILES attribute in the appropriate order.
Warning
Mandatory changes must be made or the software may not behave as expected.
IP Phones” whitepaper at www.polycom.com/support/voice/ .
3. Save the new configuration files and images (such as sip.ld) on the boot
server.
4. Reboot the phones by pressing the reboot multiple key combination.
Since the APPLICATION APP_FILE_PATH attribute of the <Ethernet
address>.cfg files references the individual sip.ld files, it is possible to
verify that an update is applied to phones of a particular model.
For example, the reference to sip.ld is changed to 2345-11605-001.sip.ld to
boot the SoundPoint IP 601 image.
The phones can be rebooted remotely through the SIP signaling protocol.
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 Provisioning <prov/>
Supporting SoundPoint IP 300 and 500 Phones
With enhancements in BootROM 4.0.0 and SIP 2.1.2, you can modify the
000000000000.cfg or <Ethernet address>.cfg configuration file to direct
phones to load the software image and configuration files based on the phone
The SIP 2.2.0 or later software distributions contain both new distribution files
for the new release and a uniquely named version of the SIP 2.1.2 release files
that is compatible with SoundPoint IP 300 and 500 phones.
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Setting up Your System
The following procedure must be used for upgrading to SIP 2.2.0 or later for
installations that have SoundPoint IP 300 and 500 phones deployed. It is also
recommended that this same approach be followed even if SoundPoint IP 300
and 500 phones are not part of the deployment as it will simplify management
of phone systems with future software releases.
To upgrade your SIP application:
1. Do one of the following steps:
a
Place the bootrom.ld file corresponding to BootROM revision 4.0.0 (or
later) onto the boot server.
b
Ensure that all phones are running BootROM 4.0.0 or later code.
2. Copy sip.ld, sip.cfg and phone1.cfg from the SIP2.2.0 or later release
distribution onto the boot server.
These are the relevant files for all phones except the SoundPoint IP 300 and
500 phones.
3. Copy sip_212.ld, sip_212.cfg, and phone1_212.cfg files from the SIP 2.2.0
or later release onto the boot server.
These are the relevant files for supporting the SoundPoint IP 300 and 500
phones.
4. Modify the 000000000000.cfg file, if required, to match your configuration
file structure.
For example:
<APPLICATION
APP_FILE_PATH="sip.ld"
APP_FILE_PATH_SPIP500="sip_212.ld"
APP_FILE_PATH_SPIP300="sip_212.ld"
CONFIG_FILES="[PHONE_MAC_ADDRESS]-user.cfg, phone1.cfg, sip.cfg"
CONFIG_FILES_SPIP500="[PHONE_MAC_ADDRESS]-user.cfg,
phone1_212.cfg, sip_212.cfg"
CONFIG_FILES_SPIP300="[PHONE_MAC_ADDRESS]-user.cfg,
phone1_212.cfg, sip_212.cfg"
MISC_FILES=""
LOG_FILE_DIRECTORY=""
OVERRIDES_DIRECTORY=""
CONTACTS_DIRECTORY=""
/>
5. Remove any <Ethernet address>.cfg files that may have been used with
earlier releases from the boot server.
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This approach takes advantage of an enhancement that was added in
Note
SIP2.0.1/BootROM 3.2.1 that allows for the substitution of the phone specific
[MACADDRESS] inside configuration files. This avoids the need to create unique
<Ethernet address>.cfg files for each phone such that the default
000000000000.cfg file can be used for all phones in a deployment.
If this approach is not used, then changes will need to be made to all the <Ethernet
address>.cfg files for SoundPoint IP 300 and 500 phones or all of the <Ethernet
address>.cfg files if it is not explicitly known which phones are SoundPoint IP 300
and 500 phones.
For more information, refer to “Technical Bulletin 35311: Supporting
SoundPoint IP 300 and IP 500 Phones with SIP 2.2 and Later Releases“ at
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4
Configuring Your System
After you set up your SoundPoint IP / SoundStation IP phones on the
network, you can allow users to place and answer calls using the default
configuration, however, you may be require some basic changes to optimize
your system for best results.
This chapter provides information for making configuration changes for:
•
•
•
•
This chapter also provides instructions on:
•
To troubleshoot any problems with your SoundPoint IP / SoundStation IP
phones on the network, refer to Troubleshooting Your SoundPoint IP /
Setting Up Basic Features
This section provides information for making configuration changes for the
following basic features:
•
•
•
•
•
•
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•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
This section also provides information for making configuration changes for
the following basic call management features:
•
•
•
•
•
•
•
•
•
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Configuring Your System
Call Log
The phone maintains a call log. The log contains call information such as
remote party identification, time and date, and call duration. It can be used to
redial previous outgoing calls, return incoming calls, and 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
when the phone is restarted.
On some SoundPoint IP platforms, missed calls and received calls appear in one
Note
list. Missed calls appear as
and received calls appear as
.
The “call list” feature can be disabled on all SoundPoint IP and SoundStation IP
platforms except the SoundPoint IP 330/320.
Configuration changes can performed centrally at the boot server:
Central
Configuration File:
sip.cfg
Enable or disable all call lists or individual call lists.
(boot server)
Call Timer
A call timer is provided on the display. A separate call timer is maintained for
each distinct call in progress. The call duration appears in hours, minutes, and
seconds.
There are no related configuration changes.
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.
Configuration changes can performed centrally at the boot server:
Central
Configuration File:
phone1.cfg
Specify the ring tone heard on an incoming call when another call is
active.
(boot server)
•
A-90.
For related configuration changes, refer to Customizable Audio Sound Effects
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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.
There are no related configuration changes.
Calling Party Identification
The phone displays the caller identity, derived from the network signalling,
when an incoming call is presented, if the information is provided by the call
server. For calls from parties for which a directory entry exists, the local name
assigned to the directory entry may optionally be substituted.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration File:
Specify whether or not to use directory name substitution.
sip.cfg
(boot server)
•
A-23.
Local
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 permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Missed Call Notification
The phone can display the number of calls missed since the user last looked at
the Missed Calls list. The types of calls that are counted as “missed” can be
configured per registration. Remote missed call notification can be used to
notify the phone when a call originally destined for it is diverted by another
entity such as a Session Initiation Protocol (SIP) server.
On some SoundPoint IP platforms, missed calls and received calls appear in one
list.
Note
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Configuring Your System
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
Turn this feature on or off.
(boot server)
sip.cfg
Configuration file:
phone1.cfg
Specify per-registration whether all missed-call events or only
remote/server-generated missed-call events will be displayed.
•
For more information, refer to Missed Call Configuration
Connected Party Identification
The identity of the remote party to which the user has connected is displayed
and logged, if the name and ID is provided by the call server. The connected
party identity is derived from the network signaling. In some cases the remote
party will be different from the called party identity due to network call
diversion.
There are no related configuration changes.
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
and phones, if using the default configuration parameters, the receive
handset/headset volume resets to nominal after each call to comply with
regulatory requirements. Handsfree volume persists with subsequent calls.
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Adjust receive and handset/headset volume.
(boot server)
•
Customizable Audio Sound Effects
Audio sound effects used for incoming call alerting and other indications are
customizable. Sound effects can be composed of patterns of synthesized tones
or sample audio files. The default sample audio files may be replaced with
alternates in .wav file format. Supported .wav formats include:
•
•
mono G.711 (13-bit dynamic range, 8-khz sample rate)
mono L16/16000 (16-bit dynamic range, 16-kHz sample rate)
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L16/16000 is not supported on SoundPoint IP 301 and SoundStation IP 4000
phones.
Note
Note
The alternate sampled audio sound effect files must be present on the boot server
or the Internet for downloading at boot time.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration File:
sip.cfg
Specify patterns used for sound effects and the individual tones or
sampled audio files used within them.
(boot server)
•
For more information, refer to Sampled Audio for Sound Effects
Local
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. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Message Waiting Indication
The phone will flash a message-waiting indicator (MWI) LED when instant
messages and voice messages are waiting.
Configuration changes can performed centrally at the boot server:
Central
(boot server)
Configuration file:
phone1.cfg
Specify per-registration whether the MWI LED is enabled or disabled.
•
For more information, refer to Message Waiting Indicator <mwi/>
Specify whether MWI notification is displayed for registration x
(pre-SIP 2.1 behavior is enabled).
•
A-23.
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).
4-9.
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Configuring Your System
Distinctive Ringing
There are three options for distinctive ringing:
1. The user can select the ring type for each line. This option has the lowest
priority.
2. The ring type for specific callers can be assigned in the contact directory.
previous section. This option has a higher priority than option 1 and a
lower priority than option 3.
3. The voIpProt.SIP.alertInfo.x.valueand
voIpProt.SIP.alertInfo.x.class fields can be used to map calls to
specific ring types. This option has the highest priority.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
Specify the mapping of Alert-Info strings to ring types.
sip.cfg
(boot server)
•
Configuration file:
phone1.cfg
Specify the ring type to be used for each line.
•
XML File: <Ethernet
address>-directory.
xml
This file can be created manually using an XML editor.
•
Local
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.
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Distinctive Call Waiting
The voIpProt.SIP.alertInfo.x.valueand
voIpProt.SIP.alertInfo.x.class fields can be used to map calls to distinct
call waiting types, currently limited to two styles.
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Specify the mapping of Alert-Info strings to call waiting types.
(boot server)
•
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Do Not Disturb
A Do Not Disturb (DND) feature is available to temporarily stop all incoming
call alerting. Calls can optionally be treated as though the phone is busy while
DND is enabled. DND can be configured as a per-registration feature.
Incoming calls received while DND is enabled are logged as missed. For more
information on forwarding calls while DND is enabled, refer to Call Forward
Server-based DND is active if the feature is enabled on both the phone and the
server and the phone is registered. The server-based DND feature is applicable
for all registrations on the phone (no per-registration mode) and it disables
local Call Forward and DND features.
Server-based DND will behave the same as per-SIP2.1 per-registration feature
with the following exceptions:
•
There is no indication on the phone’s user interface whether or not
server-based DND is active.
•
If server-based DND is enabled, but inactive, and the user presses the
DND key or selects the DND option on the Feature menu, the “Do Not
Disturb” message does not appear on the user’s phone (incoming call
alerting will continue).
Server-based DND is disabled if Shared Call Appearance or Bridged Line
Appearance is enabled.
Note
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Enable or disable server-based DND.
(boot server)
Specify whether or not DND results in incoming calls being given
busy treatment.
•
For more information, refer to Call Handling Configuration <call/>
Configuration file:
phone1.cfg
Enable or disable server-based DND as a per-registration feature.
•
Specify whether DND is treated as a per-registration feature or a
global feature on the phone.
•
A-93.
Local
Local Phone User
Interface
Enable or disable DND using the “Do Not Disturb” key on the
SoundPoint IP 301, 501, 550, 600, 601, and 650 or the “Do Not
Disturb” option on the Features menu on the SoundPoint IP 320, 330,
and 430 and SoundStation IP 4000.
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Configuring Your System
Handset, Headset, and Speakerphone
SoundPoint IP phones come standard with a handset and a dedicated
connector is provided for a headset (not supplied). The SoundPoint IP 320, 330,
430, 500, 501, 550, 600, 601, and 650 desktop phones and SoundStation IP 4000
conference phone are full-duplex speakerphones. The SoundPoint IP 301
phones is a listen-only speakerphone. The SoundPoint IP phones provide
dedicated keys for convenient selection of either the speakerphone or headset.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
Enable or disable persistent headset mode.
sip.cfg
(boot server)
•
A-23.
Local
Web Server
(if enabled)
Enable or disable persistent headset mode.
Navigate to: http://<phoneIPAddress>/coreConf.htm#us
Local Phone User
Interface
Enable or disable persistent headset mode through the Settings
menu.
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
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.
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Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Set whether the directory uses volatile storage on the phone
(required on the SoundPoint IP 500 platform for directories greater
than 25 entries).
(boot server)
•
XML file:
A sample file named 000000000000-directory~.xml (Note the extra
“~” in the filename) is included with the application file distribution.
This file can be used as a template for the per-phone <Ethernet
address>-directory.xml directories (edit contents, then rename to
<Ethernet address>-directory.xml). It also can be used to seed
new phones with an initial directory (edit contents, then remove “~”
from file name). Telephones without a local directory, such as new
units from the factory, will download the 00000000000-directory.xml
directory and base their initial directory on it. These files should be
edited with an XML editor. These files can be downloaded once per
reflash.
000000000000-direct
ory.xml
For information on file format, refer to Local Contact Directory File
Format, the following section.
Central
(boot server)
continued
XML file: <Ethernet
address>-directory.
xml
This file can be created manually using an XML editor.
For information on file format, refer to Local Contact Directory File
Format, the following section.
Local
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 Contact Directory File Format
An example of a local contact directory is shown below. The subsequent table
provides an explanation of each element.
<?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>
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...
<item>
<ln>Smith</ln>
<fn>Bill</fn>
<ct>1003</ct>
<sd>3</sd>
<rt>3</rt>
<dc/>
<ad>0</ad>
<ar>0</ar>
<bw> 0</bw>
<bb>0</bb>
</item>
</item_list>
</directory>
Element
Permitted Values
Interpretation
fn
UTF-8 encoded string
of up to 40 bytes
first name
Note: In some cases, this will be less than 40 characters due to
UTF-8’s variable length encoding.
ln
ct
UTF-8 encoded string
of up to 40 bytes
last name
UTF-8 encoded string
containing digits (the
user part of a SIP
URL) or a string that
constitutes a valid SIP
URL
contact
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.
Note: This field cannot be null or duplicated.
sd
Null, 1 to 9999
speed-dial index
Associates a particular entry with a speed dial bin for one-touch
dialing or dialing from the speed dial menu.
Note: On the SoundPoint IP 330/320, the maximum speed-dial index
is 99.
rt
Null, 1 to 21
ring type
When incoming calls can be associated with a directory entry by
matching the address fields, this field is used to specify ring type to
be used.
dc
UTF-8 encoded string
containing 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.
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Element
Permitted Values
Interpretation
ad
0,1
auto divert
If set to 1, automatically diverts callers that match the directory entry
to the address specified in divertcontact.
Note: If auto-divert is enabled, it has precedence over auto-reject.
ar
0,1
auto-reject
If set to 1, automatically rejects callers that match the directory entry.
Note: If auto-divert is also enabled, it has precedence over
auto-reject.
bw
bb
0,1
0,1
buddy watching
If set to 1, add this contact to the list of watched phones.
buddy block
If set to 1, block this contact from watching this phone.
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 Dial or Send soft key when making outgoing calls. As soon as a
digit pattern matching the digit map is found, the call setup process will
complete automatically. 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 or to replace certain matched digits (with the introduction of
“R” to the digit map).
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify impossible match behavior, trailing # behavior, digit map
matching strings, and time out value.
(boot server)
•
Configuration file:
phone1.cfg
Specify per-registration impossible match behavior, trailing #
behavior, digit map matching strings, and time out values that
override those in sip.cfg.
•
Local
Web Server
(if enabled)
Specify impossible match behavior, trailing # behavior, digit map
matching strings, and time out value.
Navigate to: http://<phoneIPAddress>/appConf.htm#ls
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
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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.
There are no related configuration changes.
Soft Key Activated User Interface
The user interface makes extensive use of intuitive, context-sensitive soft key
menus. The soft key function is shown above the key on the graphic display.
There are no related configuration changes.
Speed Dial
Entries in the local directory can be linked to the speed dial system. The speed
dial system allows calls to be placed quickly from dedicated keys as well as
from a speed dial menu.
If Presence watching is enabled for speed dial entries, their status will be
shown on the idle display (if the SIP server supports this feature). For more
Configuration changes can performed centrally at the boot server or locally:
Central
XML file:
The <sd>x</sd>element in the <Ethernet address>-directory.xml
file links a directory entry to a speed dial resource within the phone.
(boot server)
<Ethernet
Speed dial entries are mapped automatically to unused line keys (line
keys are not available on the SoundStation IP 4000 and 7000) and
are available for selection within the speed dial menu. (Press the
up-arrow key from the idle display to jump to SpeedDial).
address>-directory.
xml
•
Local
Local Phone User
Interface
The next available Speed Dial Index is assigned to new directory
entries. Key pad short cuts are available to facilitate assigning and
modifying the Speed Dial Index value for entries in the directory. The
Speed Dial Index field is used to link directory entries to speed dial
operations.
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.
Time and Date Display
The phone maintains a local clock and calendar. Time and date can be
displayed in certain operating modes such as when the phone is idle and
during a call. The clock and calendar must be synchronized to a remote Simple
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Network Time Protocol (SNTP) timeserver. The time and date displayed on
the phone will flash continuously until a successful SNTP response is received
to indicate that they are not accurate. The time and date display can use one of
several different formats and can be turned off.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
Turn time and date display on or off.
sip.cfg
(boot server)
•
A-23.
Set the time and date display formats.
•
Set the basic SNTP settings and daylight savings parameters.
•
Local
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. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Local Phone User
Interface
The basic SNTP settings can be made in the Network Configuration
menu.
The user can edit the time and date format and enable or disable the
time and date display under the Settings menu.
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. They will permanently
override global settings unless deleted through the Reset Local
Config menu selection.
Idle Display Animation
All phones except the SoundPoint IP 301 can display a customized animation
on the idle display in addition to the time and date. For example, a company
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Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
To turn idle display animation on or off.
To replace the animation used for the idle display.
(boot server)
•
•
For more information, refer to Animations <anim/> <IP_300/>,
To change the position of the idle display animation.
•
For more information, refer to Graphic Icons <gi/> <IP_300/>,
Ethernet Switch
The SoundPoint IP and SoundStation IP phones contain two Ethernet ports,
labeled LAN and PC, and an embedded Ethernet switch that runs at full
line-rate. The Ethernet switch allows a personal computer and other Ethernet
devices to connect to the office LAN by daisy chaining through the phone,
eliminating the need for a stand-alone hub. The SoundPoint IP switch gives
higher transmit priority to packets originating in the phone. The phone can be
powered through a local AC power adapter or can be line-powered (power
supplied through the signaling or idle pairs of the LAN Ethernet cable). Line
powering typically requires that the phone plugs directly into a dedicated
LAN jack. Devices that do not require LAN power can then plug into the
SoundPoint IP PC Ethernet port.
SoundPoint IP Switch - Port Priorities
To help ensure good voice quality, the Ethernet switch embedded in the
SoundPoint IP phones should be configured to give voice traffic emanating
from the phone higher transmit priority than those from a device connected to
the PC port. If not using a VLAN (VLAN set to blank in the setup menu), this
will automatically be the case. If using a VLAN, ensure that the 802.1p
priorities for both default and real-time transport protocol (RTP) packet types
are set to 2 or greater. Otherwise, these packets will compete equally with
those from the PC port. For more information, refer to Quality of Service
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Automatic Off-Hook Call Placement
The phone supports an optional automatic off-hook call placement feature for
each registration.
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
phone1.cfg
Specify which registrations have the feature and what contact to call
when going off hook.
(boot server)
•
For more information, refer to Automatic Off-Hook Call Placement
Call Hold
The purpose of hold is to pause activity on one call so that the user may use
the phone for another task, such as to make or receive another call. Network
signaling is employed to request that the remote party stop sending media and
to inform them that they are being held. A configurable local hold reminder
feature can be used to remind the user that they have placed calls on hold.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify whether RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or
a=inactive) outgoing hold signaling is used.
(boot server)
•
Specify local hold reminder options.
•
For more information, refer to Hold, Local Reminder
Local
Web Server
(if enabled)
Specify whether or not to use RFC 2543 (c=0.0.0.0) outgoing hold
signaling. The alternative is RFC 3264 (a=sendonly or a=inactive).
Navigate to: http://<phoneIPAddress>/appConf.htm#ls
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Local Phone User
Interface
Use the SIP Configuration menu to specify whether or not to use RFC
2543 (c=0.0.0.0) outgoing hold signaling. The alternative is RFC 3264
(a=sendonly or a=inactive).
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Configuring Your System
Call Transfer
Call transfer enables the user (party A) to move an existing call (party B) into
a new call between party B and another user (party C) selected by party A. The
phone offers three types of transfers:
•
Blind transfers—The call is transferred immediately to party C after party
A has finished dialing party C’s number. Party A does not hear ring-back.
•
Attended transfers—Party A dials party C’s number and hears ring-back
and decides to complete the transfer before party C answers. This option
can be disabled.
•
Consultative transfers—Party A dials party C’s number and talks
privately with party C after the call is answered, and then completes the
transfer or hangs up.
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Specify whether to allow a transfer during the proceeding state of a
consultation call.
(boot server)
•
Specify whether a transfer is blind or not.
•
For more information, refer to Call Handling Configuration <call/>
Local / Centralized Conferencing
The phone can conference together the local user with the remote parties of a
configurable number of independent calls by using the phone’s local audio
processing resources for the audio bridging. There is no dependency on
network signaling for local conferences.
The phone also supports centralized conferences for which external resources
are used such as a conference bridge. This relies on network signaling.
Conferences are not available when the G.729 codec is enabled on the
SoundStation IP 4000 conference phone.
Note
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Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Specify the conference hold behavior (all parties on hold or only host
is on hold).
(boot server)
•
For more information, refer to Call Handling Configuration <call/>
Specify whether or not all parties hear sound effects while setting up
a conference.
•
For more information, refer to Call Handling Configuration <call/>
Specify which type of conference to establish and the address of the
centralized conference resource.
•
For more information, refer to Conference Setup <conference/>
Call Forward
The phone provides a flexible call forwarding feature to forward calls to
another destination. Call forwarding can be applied in the following cases:
•
•
•
•
•
Automatically to all calls
Calls from a specific caller (extension)
When the phone is busy
When Do Not Disturb is active
After an extended period of alerting
The user can elect to manually forward calls while they are in the alerting state
to a predefined or manually specified destination. The call forwarding feature
works in conjunction with the distinctive incoming call treatment feature
to originate calls is unaffected by all call forwarding options. Each registration
has its own forwarding properties.
Server-based call forwarding is active if the feature is enabled on both the
phone and the server and the phone is registered. If server-based call
forwarding is enabled on any of the phone’s registrations, the other
registrations are not affected.
Server-based call forwarding will behave the same as per-SIP2.1 feature with
the following exceptions:
•
There is no indication on the phone’s user interface whether or not
server-based call forwarding is active.
•
If server-based call forwarding is enabled, but inactive, and the user
selects the call forward soft key, the “moving arrow” icon does not appear
on the user’s phone (incoming calls are not forwarded).
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Configuring Your System
Server-based call forwarding is disabled if Shared Call Appearance or Bridged Line
Appearance is enabled.
Note
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Enable or disable server-based call forwarding.
(boot server)
Configuration file:
phone1.cfg
Enable or disable server-based call forwarding as a per-registration
feature.
•
Set all call diversion settings including a global forward-to contact and
individual settings for call forward all, call forward busy, call forward
no-answer, and call forward do-not-disturb.
•
Local
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. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
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. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
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 changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Turn this feature on or off.
(boot server)
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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 changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Turn this feature on or off.
(boot server)
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 changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Turn this feature on or off.
(boot server)
•
Last Call Return
The phone allows server-based last call return. This feature depends on
support from a SIP server.
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Turn this feature on or off.
(boot server)
•
Specify the string sent to the server for last-call-return.
•
For more information, refer to Call Handling Configuration <call/>
Setting Up Advanced Features
This section provides information for making configuration changes for the
following advanced features:
•
•
•
•
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Configuring Your System
•
•
•
•
•
•
•
•
•
•
This section also provides information for making configuration changes for
the following advanced call server features:
•
•
•
•
•
•
Configurable Feature Keys
All key functions can be changed from the factory defaults. The scrolling
timeout for specific keys can be configured.
No feature keys on the SoundStation IP 4000 can be remapped.
Note
The rules for remapping of key functions are:
•
The phone keys that have removable key caps can be mapped to the
following:
— Any function that is implemented as a removable key cap on any of
the phones (Directories, Applications, Conference, Transfer, Redial,
Menu, Messages, Do Not Disturb, Call Lists)
— A speed-dial
— Null
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•
The phone keys without removable key caps cannot be remapped. These
include:
— Any keys on the dial pad
— Volume control
— Handsfree, Mute, Headset
— Hold
— Navigation Cluster
Configuration changes can performed centrally at the boot server:
Central
Configuration File:
sip.cfg
Set the key scrolling timeout, key functions, and sub-pointers for each
key (usually not necessary).
(boot server)
•
For more information on the default feature key layouts, refer to Default
Multiple Line Keys per Registration
More than one Line Key can be allocated to a single registration (phone
number or line). The number of Line Keys allocated per registration is
configurable.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
phone1.cfg
Specify the number of line keys to assign per registration.
(boot server)
Local
Web Server
(if enabled)
Specify the number of line keys to assign per registration.
Navigate to http://<phoneIPAddress>/reg.htm
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Local Phone User
Interface
Specify the number of line keys to assign per registration using the
SIP Configuration menu. Either the Web Server or the boot server
configuration files or the local phone user interface should be used to
configure registrations, not a mixture of these options. When the SIP
Configuration menu is used, it is assumed that all registrations use
the same server.
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Multiple Call Appearances
The phone supports multiple concurrent calls. The hold feature can be used to
pause activity on one call and switch to another call. The number of concurrent
calls per line key is configurable. Each registration can have more than one line
key assigned to it (refer to the previous section, Multiple Line Keys per
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify the default number of calls that can be active or on hold per
line key.
(boot server)
•
For more information, refer to Call Handling Configuration <call/>
Configuration file:
phone1.cfg
Specify per-registration the number of calls that can be active or on
hold per line key assigned to that registration. This will override the
default value specified in sip.cfg.
•
Local
Web Server
(if enabled)
Specify the default number of calls that can be active or on hold per
line key and the number of calls per registration that can be active or
on hold per line key assigned to that registration.
Navigate to http://<phoneIPAddress>/appConf.htm#ls and
http://<phoneIPAddress>/reg.htm
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Local Phone User
Interface
Specify per-registration the number of calls that can be active or on
hold per line key assigned to that registration using the SIP
Configuration menu. Either the Web Server or the boot server
configuration files or the local phone user interface should be used to
configure registrations, not a mixture of these options. When the SIP
Configuration menu is used, it is assumed that all registrations use
the same server.
Shared Call Appearances
Calls and lines on multiple phones can be logically related to each other. A call
that is active on one phone will be presented visually to phones that share that
call appearance. Mutual exclusion features emulate traditional PBX or key
system privacy for shared calls. Incoming calls can be presented to multiple
phones simultaneously. This feature is dependent on support from a SIP
server that binds the appearances together logically and looks after the
necessary state notifications and performs an access control function. For more
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Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
Specify whether diversion should be disabled on shared lines.
sip.cfg
(boot server)
•
A-57.
Specify line-seize subscription period.
•
Specify standard or non-standard behavior for processing line-seize
subscription for mutual exclusion feature.
•
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.
•
Specify per-registration whether diversion should be disabled on
shared lines.
•
Local
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.
Navigate to http://<phoneIPAddress>/appConf.htm#ls
Specify per-registration line type (private or shared) and line-seize
subscription period if using per-registration servers, and whether
diversion should be disabled on shared lines.
Navigate to http://<phoneIPAddress>/reg.htm
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
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.
Bridged Line Appearance
Calls and lines on multiple phones can be logically related to each other. A call
that is active on one phone will be presented visually to phones that share that
line. Mutual exclusion features emulate traditional PBX or key system privacy
for shared calls. Incoming calls can be presented to multiple phones
simultaneously. This feature is dependent on support from a SIP server that
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Configuring Your System
binds the appearances together logically and looks after the necessary state
notifications and performs an access control function. For more information,
In the configuration files, bridged lines are configured by “shared line” parameters.
Note
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify whether diversion should be disabled on shared lines.
(boot server)
•
For more information, refer to Call Handling Configuration <call/>
Configuration file:
phone1.cfg
Specify per-registration line type (private or shared) and the shared
line third party name. A shared line will subscribe to a server
providing call state information.
•
Specify per-registration whether diversion should be disabled on
shared lines.
•
Local
Web Server
(if enabled)
Specify per-registration line type (private or shared) and third party
name, and whether diversion should be disabled on shared lines.
Navigate to http://<phoneIPAddress>/reg.htm
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Local Phone User
Interface
Specify per-registration line type (private or shared) and the shared
line third party name using the SIP Configuration menu. Either the
Web Server or the boot server configuration files or the local phone
user interface should be used to configure registrations, not a mixture
of these options. When the SIP Configuration menu is used, it is
assumed that all registrations use the same server.
Busy Lamp Field
This feature is available only on SoundPoint IP 600 phones and SoundPoint IP 601
and 650 phones with an attached Expansion Module.
Note
The Busy Lamp Field (BLF) feature enhances support for a phone-based
attendant console. It allows monitoring the hook status and remote party
information of users through the busy lamp fields and displays on an
attendant console phone.
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Polycom recommends that the BLF not be used in conjunction with the Microsoft
Live Communications Server 2005 feature. For more information, refer to Microsoft
A-7).
Note
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
phone1.cfg
Specify the list SIP URI and index of the registration which will be
used to send a SUBSCRIBE to the list SIP URI specified in
(boot server)
attendant.uri
.
•
A-98.
Customizable Fonts and Indicators
The phone’s user interface can be customized by changing the fonts and
graphic icons used on the display and the LED indicator patterns. Pre-existing
fonts embedded in the software can be overwritten or new fonts can be
downloaded. The bitmaps and bitmap animations used for graphic icons on
the display can be changed and repositioned. LED flashing sequences and
colors can be changed.
Configuration changes can performed centrally at the boot server:
Central (boot
server)
Configuration File:
sip.cfg
Specify fonts to overwrite existing ones or specify new fonts.
Specify which bitmaps to use.
Specify how to create animations and LED indicator patterns.
•
•
•
Instant Messaging
The phone supports sending and receiving instant text messages. The user is
alerted to incoming messages visually and audibly. The user can view the
messages immediately or when it is convenient. For sending messages, the
user can either select a message from a preset 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
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Configuring Your System
message or by initiating a new dialog. The destination for new dialog
messages can be entered manually or selected from the contact directory, the
preferred method.
There are no related configuration changes.
Multilingual User Interface
This feature is not available on SoundPoint IP 301 phones.
Note
The system administrator or the user can select the language. Support for
major western European languages is included and additional languages can
be easily added. Support for Asian languages (Chinese, Japanese, and Korean)
is also included, but will display only on the SoundPoint IP 600, 601, and 650
and SoundStation IP 4000’s higher resolution display.
For basic character support and extended character support (available on
SoundPoint IP 600, 601, and 650 and SoundStation IP platform), refer to
characters may not be supported due to their infrequent usage.)
The multilingual feature relies on dictionary files resident on the boot server. The
dictionary files are downloaded from the boot server whenever the language is
changed or at boot time when a language other than the internal US English
language has been configured. If the dictionary files are inaccessible, the language
will revert to the internal language.
Note
Note
Currently, the multilingual feature is only available in the application. At this time,
the bootROM application is available in English only.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify the boot-up language and the selection of language choices
to be made available to the user.
(boot server)
•
For instructions on adding new languages, refer to To add new
Local
Local Phone User
Interface
The user can select the preferred language under the Settings menu.
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
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Downloadable Fonts
New fonts can be loaded onto the phone. For guidelines on downloading
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 during the life cycle of a call. These call progress tones are easily
configurable for compatibility with worldwide telephony standards or local
preferences.
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Specify the basic tone frequencies, levels, and basic repetitive
cadences.
(boot server)
•
Specify downloaded sampled audio files for advanced call progress
tones.
•
For more information, refer to Sampled Audio for Sound Effects
Specify patterns.
•
Microbrowser
The SoundPoint IP 430, 501, 550, 600, 601, and 650 phones and the
SoundStation IP 4000 phone supports an XHTML Microbrowser. This can be
launched by pressing the Applications key, or if there isn’t one on the phone,
it can be accessed through the Menu key by selecting Features, and then
Applications.
As of SIP 2.2.0, the Services key and menu entry are renamed Applications,
however the functionality remains the same.
Note
Two instances of the Microbrowser may run concurrently:
•
•
An instance with standard interactive user interface
An instance that does not support user input, but appears in a window on
the idle display
For more information, refer to the Microbrowser Developers’s Guide.
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Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify the Application browser home page, a proxy to use, and size
limits.
(boot server)
•
Local
Web Server
(if enabled)
Specify the Applications browser home page and proxy to use.
Navigate to http://<phoneIPAddress>/coreConf.htm#mb
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Real-Time Transport Protocol Ports
The phone is compatible with RFC 1889 - RTP: A Transport Protocol for
Real-Time Applications - and the updated RFCs 3550 and 3551. Consistent
with RFC 1889, the phone treats all RTP streams as bi-directional from a
control perspective and expects that both RTP end points will negotiate the
respective destination IP addresses and ports. This allows real-time transport
control protocol (RTCP) to operate correctly even with RTP media flowing in
only a single direction, or not at all. It also allows greater security: packets from
unauthorized sources can be rejected.
The phone can filter incoming RTP packets arriving on a particular port by IP
address. Packets arriving from a non-negotiated IP address can be discarded.
The phone can also enforce symmetric port operation for RTP packets: packets
arriving with the source port set to other than the negotiated remote sink port
can be rejected.
The phone can also jam the destination transport port to a specified value
regardless of the negotiated port. This can be useful for punching through
firewalls. When this is enabled, all RTP traffic will be sent to the specified port
and will be expected to arrive on that port as well. Incoming packets are sorted
by the source IP address and port, allowing multiple RTP streams to be
multiplexed.
The RTP port range used by the phone can be specified. Since conferencing
and multiple RTP streams are supported, several ports can be used
concurrently. Consistent with RFC 1889, the next higher odd port is used to
send and receive RTCP.
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Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify whether to filter incoming RTP packets by IP address,
whether to require symmetric port usage, whether to jam the
destination port and specify the local RTP port range start.
(boot server)
•
Local
Web Server
(if enabled)
Specify whether to filter incoming RTP packets by IP address,
whether to require symmetric port usage, whether to jam the
destination port and specify the local RTP port range start.
Navigate to: http://<phoneIPAddress>/netConf.htm#rt
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Network Address Translation
The phone can work with certain types of network address translation (NAT).
The phone’s signaling and RTP traffic use symmetric ports (the source port in
transmitted packets is the same as the associated listening port used to receive
packets) and the external IP address and ports used by the NAT on the phone’s
behalf can be configured on a per-phone basis.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify the external NAT IP address and the ports to be used for
signaling and RTP traffic.
(boot server)
•
For more information, refer to Network Address Translation
Local
Web Server
(if enabled)
Specify the external NAT IP address and the ports to be used for
signaling and the RTP traffic.
Navigate to: http://<phoneIPAddress>/netConf.htm#na
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Voice Mail Integration
The phone is compatible with voice mail servers. The subscribe contact and
callback mode can be configured per user/registration on the phone. The
phone can be configured with a SIP URL to be called automatically by the
phone when the user elects to retrieve messages. Voice mail access can be
configured to be through a single key press (for example, the Messages key on
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Configuring Your System
the SoundPoint IP 430, 500, 501, 550, 600, 601, and 650). A message-waiting
signal from a voice mail server will trigger the message-waiting indicator to
flash.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
For one-touch voice mail access, enable the “one-touch voice mail”
user preference.
(boot server)
•
A-23.
Configuration file:
phone1.cfg
For one-touch voice mail access, bypass instant messages to
remove the step of selecting between instant messages and voice
mail after pressing the Messages key on the SoundPoint IP 430, 500,
501, 550, 600, 601, and 650 (instant messages are still accessible
from the Main Menu).
On a per-registration basis, specify a subscribe contact for solicited
NOTIFY applications, a callback mode (self call-back or another
contact), and the contact to call when the user accesses voice mail.
•
Local
Web Server
(if enabled)
For one-touch voice mail access, enable the “one-touch voice mail”
user preference and bypass instant messages to remove the step of
selecting between instant messages and voice mail after pressing the
Messages key on the SoundPoint IP 430, 500, 501, 550, 600, 601,
and 650 (instant messages are still accessible from the Main Menu).
Navigate to http://<phoneIPAddress>/coreConf.htm#us
On a per-registration basis, specify a subscribe contact for solicited
NOTIFY applications, a callback mode (self call-back or another
contact) to call when the user accesses voice mail.
Navigate to http://<phoneIPAddress>/reg.htm
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Multiple Registrations
The SoundPoint IP 301, 320, 330, and 430 support a maximum of two
registrations, the SoundPoint IP 501 supports three, the SoundPoint IP 550
supports four, and the SoundPoint IP 600, 601, and 650 support 6. Up to three
SoundPoint IP Expansion Modules can be added to a single host SoundPoint
IP 601 and 650 phone increasing the total number of buttons to 12 registrations
on the IP 601 and 34 registrations on the IP 650. The SoundStation IP 4000
supports a single registration.
Each registration can be mapped to one or more line keys (a line key can be
used for only one registration). The user can select which registration to use for
outgoing calls or which to use when initiating new instant message dialogs.
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Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify the local SIP signaling port and an array of SIP servers to
register to. For each server specify the registration period and the
signaling failure behavior.
(boot server)
•
Configuration file:
phone1.cfg
For up to twelve registrations, specify a display name, a SIP address,
an optional display label, an authentication user ID and password, the
number of line keys to use, and an optional array of registration
servers. The authentication user ID and password are optional and
for security reasons can be omitted from the configuration files. The
local flash parameters will be used instead. The optional array of
servers and their associated parameters will override the servers
specified in sip.cfg if non-Null.
•
Local
Web Server
(if enabled)
Specify the local SIP signaling port and an array of SIP servers to
register to.
Navigate to http://<phoneIPAddress>/appConf.htm#se
For up to six registrations (depending on the phone model, in this
case the maximum is six even for the IP 601 and 650), specify a
display name, a SIP address, an optional display label, an
authentication user ID and password, the number of line keys to use,
and an optional array of registration servers. The authentication user
ID and password are optional and for security reasons can be omitted
from the configuration files. The local flash parameters will be used
instead. The optional array of servers will override the servers
specified in sip.cfg in non-Null. This will also override the servers on
the appConf.htm web page.
Navigate to http://<phoneIPAddress>/reg.htm
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
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Configuring Your System
Local
(continued)
Local Phone User
Interface
Use the SIP Configuration menu to specify the local SIP signaling
port, a default SIP server to register to and registration information for
up to twelve registrations (depending on the phone model). The SIP
Configuration menu contains a sub-set of all the parameters available
in the configuration files.
Either the Web Server or the boot server configuration files or the
local phone user interface should be used to configure registrations,
not a mixture of these options. When the SIP Configuration menu is
used, it is assumed that all registrations use the same server.
Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Automatic Call Distribution
The phone allows automatic call distribution (ACD) login and logout. This
feature depends on support from a SIP server.
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Turn this feature on or off.
(boot server)
Configuration file:
phone1.cfg
Enable this feature per registration.
•
The phone also supports ACD agent available and unavailable. This feature
depends on support from a SIP server.
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Turn this feature on or off.
(boot server)
Configuration file:
phone1.cfg
Enable this feature per registration.
•
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Server Redundancy
Server redundancy is often required in VoIP deployments to ensure continuity
of phone service for events where the call server needs to be taken offline for
maintenance, the server fails, or the connection from the phone to the server
fails.
Two types of redundancy are possible:
•
Fail-over: In this mode, the full phone system functionality is preserved by
having a second equivalent capability call server take over from the one
that has gone down/off-line. This mode of operation should be done
using DNS mechanisms or “IP Address Moving” from the primary to the
back-up server.
•
Fallback: In this mode, a second less featured call server (router or
gateway device) with SIP capability takes over call control to provide basic
calling capability, but without some of the richer features offered by the
primary call server (for example, shared lines, presence, and Message
Waiting Indicator). Polycom phones support configuration of multiple
servers per SIP registration for this purpose.
In some cases, a combination of the two may be deployed.
Your SIP server provider should be consulted for recommended methods of
configuring phones and servers for fail-over configuration.
Note
supported. Customers that are using the reg.x.server.y. configuration
Warning
parameters where y>=2 should take care to ensure that their current deployments
are not adversely affected. For example the phone will only support advanced SIP
features such as shared lines, missed calls, presence with the primary server (y=1).
For more information, refer to “Technical Bulletin 5844: SIP Server Fallback
Enhancements on SoundPoint IP Phones” at
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Specify global primary and fallback server configuration parameters.
(boot server)
Configuration file:
Specify per registration primary and fallback server configuration
phone1.cfg
parameters values that override those in sip.cfg.
•
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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. 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.
Failure to resolve a DNS name is treated as signalling failure that will cause a
failover.
Note
Behavior When the Primary Server Connection Fails
For Outgoing Calls (INVITE Fallback)
When the user initiates a call, the phone will go through the following steps to
connect the call:
1. Try to make the call using the working server.
2. If the working server does not respond correctly to the INVITE, then try
and make a call using the next server in the list (even if there is no current
registration with these servers). This could be the case if the Internet
connection has gone down, but the registration to the working server has
not yet expired.
3. If the second server is also unavailable, the phone will try all possible
servers (even those not currently registered) until it either succeeds in
making a call or exhausts the list at which point the call will fail.
At the start of a call, server availability is determined by SIP signaling failure.
SIP signaling failure depends on the SIP protocol being used as described
below:
•
If TCP is used, then the signaling fails if the connection fails or the Send
fails.
•
If UDP is used, then the signaling fails if ICMP is detected or if the signal
times out. If the signaling has been attempted through all servers in the list
and this is the last server, then the signaling fails after the complete UDP
timeout defined in RFC 3261. If it is not the last server in the list, the
maximum number of retries using the configurable retry timeout is used.
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If DNS is used to resolve the address for Servers, the DNS server is unavailable,
Warning
and the TTL for the DNS records has expired, the phone will attempt to contact the
DNS server to resolve the address of all servers in its list before initiating a call.
These attempts will timeout, but the timeout mechanism can cause long delays (for
example, two minutes) before the phone call proceeds “using the working server”.
To mitigate this issue, long TTLs should be used. It is strongly recommended that
an on-site DNS server is deployed as part of the redundancy solution.
Hosted VoIP Service
Provider
Call Server 1B
Call Server 1A
Internet
DNS Server
VoIP SMB Customer
Premise
SIP Capable Router
Server2
`
`
PSTN
PSTN Gateway
`
Phone Configuration
The phones at the customer site are configured as follows:
•
Server 1 (the primary server) will be configured with the address of the
service provider call server. The IP address of the server(s) to be used will
be provided by the DNS server. For example:
reg.1.server.1.address="voipserver.serviceprovider.com"
•
Server 2 (the fallback server) will be configured to the address of the
router/gateway that provides the fallback telephony support and is
on-site. For example:
reg.1.server.2.address=172.23.0.1
It is possible to configure the phone for more than two servers per registration, but
you need to exercise caution when doing this to ensure that the phone and network
load generated by registration refresh of multiple registrations do not become
excessive. This would be of particularly concern if a phone had multiple
registrations with multiple servers per registration and it is expected that some of
these servers will be unavailable.
Note
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Configuring Your System
Phone Operation for Registration
After the phone has booted up, it will register to all the servers that are
configured.
Server 1 is the primary server and supports greater SIP functionality than any
of servers. For example, SUBSCRIBE/NOTIFY services (used for features such
as shared lines, presence, and BLF) will only be established with Server 1.
Upon registration timer expiry of each server registration, the phone will
attempt to re-register. If this is unsuccessful, normal SIP re-registration
behavior (typically at intervals of 30 to 60 seconds) will proceed and continue
until the registration is successful (for example, when the Internet link is once
again operational). While the primary server registration is unavailable, the
next highest priority server in the list will serve as the working server. As soon
as the primary server registration succeeds, it will return to being the working
server.
If reg.x.server.y.registeris set to 0, then phone will not register to that server.
However, the INVITE will fail over to that server if all higher priority servers are
down.
Note
Recommended Practices for Fallback Deployments
In situations where server redundancy for fall-back purpose is used, the
following measures should be taken to optimize the effectiveness of the
solution:
1. Deploy an on-site DNS server to avoid long call initiation delays that can
result if the DNS server records expire.
2. Do not use OutBoundProxy configurations on the phone if the
OutBoundProxy could be unreachable when the fallback occurs.
SoundPoint IP phones can only be configured with one OutBoundProxy
per registration and all traffic for that registration will be routed through
this proxy for all servers attached to that registration. If Server 2 is not
accessible through the configured proxy, call signaling with Server 2 will
fail.
3. Avoid using too many servers as part of the redundancy configuration as
each registration will generate more traffic.
4. Educate users as to the features that will not be available when in
“fallback” operating mode.
Presence
The Presence feature allows the phone to monitor the status of other
users/devices and allows other users to monitor it. The status of monitored
users is displayed visually and is updated in real time in the Buddies display
screen or, for speed dial entries, on the phone’s idle display. Users can block
others from monitoring their phones and are notified when a change in
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monitored status occurs. Phone status changes are broadcast automatically to
monitoring phones when the user engages in calls or invokes do-not-disturb.
The user can also manually specify a state to convey, overriding, and perhaps
masking, the automatic behavior.
Notification when a change in monitored status occurs will be available in a
subsequent release.
Note
The presence feature works differently when Microsoft Live Communications
Server 2005 is used as the call server. For more information, refer to the
Configuration changes can performed centrally at the boot server:
Central
XML file: <Ethernet
address>-directory.
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.
(boot server)
•
Local
Local Phone User
Interface
The user can edit the directory contents. The Watch Buddy and
Block Buddy fields control the buddy behavior of contacts.
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.
Microsoft Live Communications Server 2005 Integration
SoundPoint IP phones can used with Microsoft Live Communications
Server 2005 and Microsoft Office Communicator to help improve business
efficiencies and increase productivity and to share ideas and information
immediately with business contacts.
Any contacts added through the SoundPoint IP phone’s buddy list will appear in as
a contact in Microsoft Office Communicator and Windows Messenger.
Note
Polycom recommends that the BLF not be used in conjunction with the Microsoft
Live Communications Server 2005 feature. For more information, refer to Busy
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Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Specify that support for Microsoft Live Communications Server 2005
is enabled.
(boot server)
•
Specify the line/registration number used to send SUBSCRIBE for
presence.
•
Turn the presence and messaging features on or off.
•
Configuration file:
phone1.cfg
Specify the number of line keys to assign per registration.
•
Specify the line/registration number which has roaming buddies
support enabled.
•
For more information, refer to Roaming Buddies
Specify the line/registration number which has roaming privacy
support enabled.
•
For more information, refer to Roaming Privacy
Configuration File Example
SoundPoint IP phones can be deployed in two basic methods. In the first
method, Microsoft Live Communications Server 2005 serves as the call server
and the phones have a single registration. In the second method, the phone has
a primary registration to call server—that is not Microsoft Live
Communications Server (LCS)—and a secondary registration to LCS for
presence purposes.
To set up a single registration with Microsoft Live Communications Server 2005
as the call server:
1. Modify the sip.cfg configuration file as follows:
a
b
c
Open sip.cfg in an XML editor.
Locate the feature parameter.
For the feature.1.name= presenceattribute, set
feature.1.enabledto 1.
d
For the feature.2.name = messagingattribute, set
feature.2.enabledto 1.
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e
Locate the voIpProtparameter.
Set the voIpProt.server.x.transportattribute to TCPpreferred or
TLS.
Your selection depends on the LCS configuration.
The TLS protocol is not supported on SoundPoint IP 300 and 500 phones.
Note
f
Set the voIpProt.server.x.addressto the LCS address.
For example, voIpProt.server.1.address = "lcs2005.local"
Set the voIpProt.SIP.lcsattribute to 1.
g
h
(Optional) If SIP forking is desired, set voIpProt.SIP.ms-forking
attribute to 1.
i
Save the modified sip.cfg configuration file.
2. Modify the phone1.cfg configuration file as follows:
a
b
c
Open phone1.cfg in an XML editor.
Locate the registration parameter.
Set the reg.1.addressto the LCS address.
For example, reg.1.address = "7778"
Set the reg.1.server.y.addressto the LCS server name.
d
e
(Optional) Set the reg.1.server.y.transportattribute to
TCPpreferred or TLS.
Your selection depends on the LCS configuration.
Set reg.1.auth.userIdto the phone's LCS username.
For example, reg.1.auth.userId = "jbloggs"
Set reg.1.auth.passwordto the LCS password.
For example, reg.1.auth.password = "Password2"
Locate the roaming_buddiesattribute.
f
g
h
i
Set the roaming_buddies.regelement to 1.
Locate the roaming_privacyattribute.
j
k
Set the roaming_privacy.regelement to 1.
Save the modified phone1.cfg configuration file.
l
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To set up a dual registration with Microsoft Live Communications Server 2005 as
the presence server:
1. (Optional) Modify the sip.cfg configuration file as follows:
a
b
c
Open sip.cfg in an XML editor.
Locate the feature parameter.
For the feature.1.name = presenceattribute, set
feature.1.enabledto 1.
d
For the feature.2.name = messagingattribute, set
feature.2.enabledto 1.
e
f
Locate the voIpProtparameter.
If SIP forking is desired, set voIpProt.SIP.ms-forkingattribute to 1.
g
Save the modified sip.cfg configuration file.
2. Modify the phone1.cfg configuration file as follows:
a
b
c
Open phone1.cfg in an XML editor.
Locate the registration parameter.
Select a registration to be used for the Microsoft Live Communications
Server 2005.
Typically, this would be 2.
d
Set the reg.x.addressto the LCS address.
For example, reg.2.address = "7778"
Set the reg.x.server.y.addressto the LCS server name.
e
f
(Optional) Set the reg.2.server.y.transportattribute to
TCPpreferred or TLS.
Your selection depends on the LCS configuration.
Set reg.x.auth.userIdto the phone's LCS username.
For example, reg.2.auth.userId = "jbloggs"
Set reg.x.auth.passwordto the LCS password.
For example, reg.2.auth.password = "Password2"
Locate the roaming_buddiesattribute.
g
h
i
j
Set the roaming_buddies.regelement to the number corresponding
to the LCS registration.
For example, roaming_buddies.reg = 2
Locate the roaming_privacyattribute.
k
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l
Set the roaming_privacy.regelement to the number corresponding
to the LCS registration.
For example, roaming_privacy.reg = 2
Save the modified phone1.cfg configuration file.
m
Setting Up Audio Features
Proprietary state-of-the-art digital signal processing (DSP) technology is used
to provide an excellent audio experience.
This section provides information for making configuration changes for the
following audio-related features:
•
•
•
•
•
•
•
•
•
•
•
•
Low-Delay Audio Packet Transmission
The phone is designed to minimize latency for audio packet transmission.
There are no related configuration changes.
Jitter Buffer and Packet Error Concealment
The phone employs a high-performance jitter buffer and packet error
concealment system designed to mitigate packet inter-arrival jitter and
out-of-order or lost (lost or excessively delayed by the network) packets. The
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Configuring Your System
jitter buffer is adaptive and configurable for different network environments.
When packets are lost, a concealment algorithm minimizes the resulting
negative audio consequences.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Set the jitter buffer tuning parameters including minimum and
maximum size and shrink aggression.
(boot server)
•
Local
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 permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
Voice Activity Detection
The purpose of voice activity detection (VAD) is to conserve network
bandwidth by detecting periods of relative “silence” in the transmit data path
and replacing that silence efficiently with special packets that indicate silence
is occurring. For those compression algorithms without an inherent VAD
function, such as G.711, the phone is compatible with the comprehensive
codec-independent comfort noise transmission algorithm specified in RFC
3389. This algorithm is derived from G.711 Appendix II, which defines a
comfort noise (CN) payload format (or bit-stream) for G.711 use in
packet-based, multimedia communication systems. The phone generates CN
packets (also known as Silence Insertion Descriptor (SID) frames) and also
decodes CN packets, efficiently regenerating a facsimile of the background
noise at the remote end.
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Enable or disable VAD and set the detection threshold.
(boot server)
•
DTMF Tone Generation
The phone generates dual tone multi-frequency (DTMF) tones in response to
user dialing on the dial pad. These tones are transmitted in the real-time
transport protocol (RTP) streams of connected calls. The phone can encode the
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Administrator’s Guide SoundPoint IP / SoundStation IP
DTMF tones using the active voice codec or using RFC 2833 compatible
encoding. The coding format decision is based on the capabilities of the remote
end point.
Configuration changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Set the DTMF tone levels, autodialing on and off times, and other
parameters.
(boot server)
•
For more information, refer to Dual Tone Multi-Frequency
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 technique 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 changes can performed centrally at the boot server:
Central
Configuration file:
sip.cfg
Enable or disable RFC 2833 support in SDP offers and specify the
payload value to use in SDP offers.
(boot server)
•
For more information, refer to Dual Tone Multi-Frequency
Acoustic Echo Cancellation
The phone employs advanced acoustic echo cancellation (AEC) for hands-free
operation. Both linear and non-linear techniques are employed to aggressively
reduce echo yet provide for natural full-duplex communication patterns.
When using the handset on any SoundPoint IP phones, AEC is not normally
required. In certain situations, where echo is experienced by the far-end party,
when the user is on the handset, AEC may be enabled to reduce/avoid this
echo. To achieve this, make the following changes in the sip.cfg configuration
file (default settings for these parameters are disabled):
voice.aec.hs.enable = 1
voice.aes.hs.enable = 1
voice.ns.hs.enable = 1
voice.ns.hs.signalAttn = -6
voice.ns.hs.silenceAttn = -9
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Configuring Your System
For the SoundPoint IP 501 and 601, utilizing acoustic echo cancellation will
introduce a small delay increase into the audio path which might cause a lower
voice quality.
AEC on the SoundPoint IP 301 handset is not supported.
Note
Audio Codecs
The following table summarizes the phone’s audio codec support:
Effective
Sample
Rate
audio
bandwidth
Algorithm
G.711μ-law
G.711a-law
G.722
MIME Type
PMCU
Ref.
Bit Rate
64 Kbps
64 Kbps
64 Kbps
Frame Size
10ms - 80ms
10ms - 80ms
10ms - 80ms
20ms - 80ms
RFC 1890
RFC 1890
RFC 1890
RFC 3047
8 Ksps
8 Ksps
16 Ksps
16 Ksps
3.5KHz
3.5KHz
7 KHz
PCMA
G722/8000
G722/16000
G.722.1
16 Kbps,
24 Kbps,
32 Kbps
7 KHz
G.729AB
SID
G729
RFC 1890
RFC 3389
RFC 2833
8 Kbps
N/A
8 Ksps
N/A
10ms - 80ms
N/A
3.5KHz
N/A
CN
RFC 2833
phone-event
N/A
N/A
N/A
N/A
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify codec priority, preferred payload sizes, and jitter buffer tuning
parameters.
(boot server)
•
For more information, refer to Codec Preferences <codecPref/>
Local
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 permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.
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Background Noise Suppression
Background noise suppression (BNS) is designed primarily for hands-free
operation and reduces background noise to enhance communication in noisy
environments.
There are no related configuration changes.
Comfort Noise Fill
Comfort noise fill is designed to help provide a consistent noise level to the
remote user of a hands-free call. Fluctuations in perceived background noise
levels are an undesirable side effect of the non-linear component of most AEC
systems. This feature uses noise synthesis techniques to smooth out the noise
level in the direction toward the remote user, providing a more natural call
experience.
There are no related configuration changes.
Automatic Gain Control
Automatic Gain Control (AGC) is applicable to hands-free operation and is
used to boost the transmit gain of the local talker in certain circumstances. This
increases the effective user-phone radius and helps with the intelligibility of
soft-talkers.
There are no related configuration changes.
IP Type-of-Service
The “type of service” field in an IP packet header consists of four
type-of-service (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.
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify protocol-specific IP TOS settings.
(boot server)
Local
Web Server
(if enabled)
Specify IP TOS settings.
Navigate to: http://<phoneIPAddress>/netConf.htm#qo
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Configuring Your System
IEEE 802.1p/Q
The phone will tag all Ethernet packets it transmits with an 802.1Q VLAN
header for one of the following reasons:
•
•
When it has a valid VLAN ID set in its network configuration
When it is instructed to tag packets through Cisco Discovery Protocol
(CDP) running on a connected Ethernet switch
•
3-7)
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 changes can performed centrally at the boot server or locally:
Central
Configuration file:
Specify default and protocol-specific 802.1p/Q settings.
sip.cfg
(boot server)
•
For more information, refer to Ethernet IEEE 802.1p/Q
Local
Web Server
(if enabled)
Specify 802.1p/Q settings.
Navigate to http://<phoneIPAddress>/netConf.htm#qo
Local Phone User
Interface
Specify whether CDP is to be used or manually set the VLAN ID or
configure DHCP VLAN Discovery.
Phase 1: bootRom - Navigate to: SETUP menu during auto-boot
countdown.
Phase 2: Application - Navigate to:
Menu>Settings>Advanced>Admin Settings>Network Configuration
•
3-2.
Setting Up Security Features
This section provides information for making configuration changes for the
following security-related features:
•
•
•
•
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Local User and Administrator Privilege Levels
Several local settings menus are protected with two privilege levels, user and
administrator, each with its own password. The phone will prompt for either
the user or administrator password before granting access to the various menu
options. When the user password is requested, the administrator password
will also work. The web server is protected by the administrator password
4-50).
Configuration changes can performed centrally at the boot server or locally:
Central
Configuration file:
sip.cfg
Specify the minimum lengths for the user and administrator
passwords.
(boot server)
•
For more information, refer to Password Lengths
Local
Web Server
(if enabled)
None.
Local Phone User
Interface
The user and administrator passwords can be changed under the
Settings menu or through configuration parameters (refer to Flash
ASCII characters 32-127 (0x20-0x7F) only.
Changes are saved to local flash but are not backed up to <Ethernet
address>-phone.cfg on the boot server for security reasons.
Custom Certificates
The phone trusts certificates issued by widely recognized certificate
authorities when trying to establish a connection to a boot server for
C-1.
In addition, custom certificates can be added to the phone. This is done by
using the SSL Security menu on the phone to provide the URL of the custom
certificate then select an option to use this custom certificate.
For more information on using custom certificates, refer to “Technical Bulletin
17877: Using Custom Certificates With SoundPoint IP Phones” at
www.polycom.com/support/voice/ .
Note
Configuration changes can performed locally:
Local
Local Phone User
Interface
The custom certificate can be specified and the type of certificate to
trust can be set under the Settings menu.
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Configuring Your System
Incoming Signaling Validation
The three optional levels of security for validating incoming network signaling
are:
•
•
•
Source IP address validation
Digest authentication
Source IP address validation and digest authentication
Configuration changes can performed centrally at the boot server:
Central
Configuration File:
sip.cfg
Specify the type of validation to perform on a request-by-request
basis, appropriate to specific event types in some cases.
(boot server)
•
For more information, refer to Request Validation
Configuration File Encryption
Configuration files (excluding the master configuration file), contact
directories, and configuration override files can all be encrypted.
The SoundPoint IP 300 and 500 phones will always fail at decrypting files. These
phones will recognize that a file is encrypted, but cannot decrypt it and will display
an error. Encrypted configuration files can only be decrypted on the SoundPoint IP
301, 320, 330, 430, 501,550, 600, 601, and 650 and the SoundStation IP 4000
phones.
Note
The master configuration file cannot be encrypted on the boot server. This file is
downloaded by the bootROM that does not recognize encrypted files. For more
For more information on encrypting configuration files including determining
whether an encrypted file is the same as an unencrypted file and using the
Configuration changes can performed centrally at the boot server:
Central
(boot server)
Configuration File:
sip.cfg
Specify the phone-specific contact directory and the
phone-specific configuration override file.
•
For more information, refer to Encryption <encryption/>
Configuration file:
Change the encryption key.
<device>.cfg
•
For more information, refer to refer to Flash Parameter
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Configuring SoundPoint IP / SoundStation IP Phones Locally
A local phone-based configuration web server is available, unless it is disabled
through sip.cfg. It can be used as the only method of modifying phone
configuration or as a distributed method of augmenting a centralized
The phone’s local user interface also permits many application settings to be
modified, such as SIP server address, ring type, or regional settings such as
time/date format and language.
Local Web
Point your web browser to http://<phoneIPAddress>/.
Server Access
Configuration pages are accessible from the menu along the top banner.
The web server will issue an authentication challenge to all pages except for
the home page.
Credentials are (case sensitive):
User Name: Polycom
Password: The administrator password is used for this.
Local Settings
Menu Access
Some items in the Settings menu are locked to prevent accidental changes.
To unlock these menus, enter the user or administrator passwords.
The administrator password can be used anywhere that the user password is
used.
Factory default passwords are:
User password: 123
Administrator password: 456
Passwords:
Administrator
password
required.
Network Configuration
SIP Configuration
SSL Security settings
Reset to Default - local configuration, device settings, and file system format
User password
required.
Restart Phone
Changes made through the web server or local user interface are stored
internally as overrides. These overrides take precedence over settings
contained in the configuration obtained from the boot server.
If the boot server permits uploads, these override setting will be saved in a file
called <Ethernet address>-phone.cfg on the boot server as well in flash
memory.
Local configuration changes will continue to override the boot server-derived
configuration until deleted through the Reset Local Config menu selection.
Warning
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5
Troubleshooting Your SoundPoint IP
/ SoundStation IP Phones
This chapter provides you with some tools and techniques for troubleshooting
SoundPoint IP / SoundStation IP phones and installations. The phone can
provide feedback in the form of on-screen error messages, status indicators,
and log files for troubleshooting issues.
This chapter includes information on:
•
•
•
•
This chapter also presents phone issues, likely causes, and corrective actions.
Issues are grouped as follows:
•
•
•
•
•
•
•
Review the latest Release Notes for the SIP application for known problems and
possible workarounds. For the latest Release Notes and the latest version of this
Administrator’s Guide, go to Polycom Technical Support at
If your problems is not listed in this chapter nor described in the latest Release
Notes, contact your Certified Polycom Reseller for support.
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Error Messages
There are several different error messages that can be displayed on the phone
when it is booting. Some of these errors are fatal, meaning that the phone will
not able to boot until this issue has been resolved, and some are recoverable,
meaning the phone will continue booting after the error, but the configuration
of the phone may not be what you were expecting.
BootROM Error Messages
Most of these errors are also logged on the phone’s boot log, however, if you
are having trouble connecting to the boot server, the phone will likely not be
able to upload the boot log for you to examine.
Failed to get boot parameters via DHCP
The phone does not have an IP address and therefore cannot boot. Check that
all cables are connected, the DHCP server is running and that the phone has
not been put into a VLAN which is different from the DHCP server. Check the
DHCP configuration.
Application <file name> is not compatible with this phone!
When the bootROM displays an error like “The application is not compatible”,
it means an application file was downloaded from the boot server, but it
cannot be installed on this phone. This issue can usually be resolved by finding
a software image that is compatible with the hardware or the bootROM being
used and installing this on the boot server. There are various different
hardware and software dependencies. Refer to the latest Release Notes for
details on the version you are using.
Could not contact boot server, using existing configuration
The phone could not contact the boot server, but the causes may be numerous.
It may be cabling issue, it may be related to DHCP configuration, or it could
be a problem with the boot server itself. The phone can recover from this error
so long as it previously downloaded a valid application bootROM image and
all of the necessary configuration files.
Error, application is not present!
There is no application stored in flash memory and the phone cannot boot. A
compatible SIP application must be downloaded into the phone using one of
the supported provisioning protocols. You need to resolve the issue of
connecting to the boot server. This error is typically a result one of the above
errors. This error is fatal.
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Troubleshooting Your SoundPoint IP / SoundStation IP Phones
Not all configuration files were present on the server
Similarly, a message about configuration files not being present, means that
the phone was able to reach the boot server, but that it was not able to find all
the necessary files. So long as the files exist in flash memory, the phone can
boot following this error.
This error does not occur with the current BootROM.
Note
Error loading <file name>
When the required file does not exist in flash memory and cannot be found on
the boot server, the “Error loading” message will tell you which file could not
be found. This error only remains on the screen for a few seconds so you need
to watch closely. The phone reboots.
This error does not occur with the current BootROM.
Note
Application Error Messages
Config file error. Error is <Hex #>
If there is an error in the configuration file, you will not be able to reboot the
phones. You must review the boot server configuration, make the correction,
and reapply the configuration file by restarting the phones.
Network link is down
Since the SoundPoint IP / SoundStation IP phones do not have an LED
indicating network LINK status like many networking devices, if a link failure
is detected while the phone is running a message saying “Network link is
down” will be displayed. This message will be shown on the screen whenever
the phone is not in the menu system and will remain on screen until the link
problem is resolved.
Status
When the phone is unable to register with the call control server, the icon for
that will be shown
shown
. Once the phone is able to register, the icon will be
.
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Blinking Time
If the phone has not been able to contact the SNTP server or if one has not been
configured, the date/time display will flash until this is fixed. If an SNTP is not
available, the data/time display can be turned off so that the flashing display
is not a distraction.
Status Menu
Debugging of single phone may be possible through an examination of the
phone’s status menu. Press Menu, select Status, and then press the Select soft
key.
Under the Platform selection, you can get details on the phone’s serial number
or MAC address, the current IP address, the bootROM version, the application
version, the name of the configuration files in use, and the address of the boot
server.
In the Network menu, the phone will provide information about TCP/IP
setting, Ethernet port speed, connectivity status of the PC port, and statistics
on packets sent and received since last boot. This would also be a good place
to look and see how long it’s been since the phone rebooted. The Call Statistics
screen shows packets sent and received on the last call.
The Lines menu will give you details about the status of each line that has been
configured on the phone.
Finally, the Diagnostics menu offers a series of hardware tests to verify correct
operation of the microphone, speaker, handset, and third party headset, if
present. It will also let you test that each of the keys on the phone is working,
and it will display the function that has been assigned to each of the keys in the
configuration. This is also where you can test the LCD for faulty pixels.
In addition to the hardware tests, the Diagnostics menu has a series of
real-time graphs for CPU, network and memory utilization that can be helpful
in diagnosing performance issues.
Log Files
SoundPoint IP and SoundStation IP phones will log various events to files
stored in the flash file system and will periodically upload these log files to the
boot server. The files are stored in the phone’s home directory or a
user-configurable directory.
There is one log file for the bootROM and one for the application. When a
phone uploads its log files, they are saved on the boot server with the MAC
address of the phone prepended to the file name. For example,
00f4f200360b-boot.log and 00f4f200360b-app.log are the files associated with
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Troubleshooting Your SoundPoint IP / SoundStation IP Phones
MAC address 00f4f200360b. The bootROM log file is uploaded to the boot
server after every reboot. The application log file is uploaded periodically or
when the local copy reaches a predetermined size.
Both log files can be uploaded on demand using a multiple key combination
files, namely, mac-boot.log, app-boot.log, mac-now-boot.log, and
mac-now-app.log. The “now_” logs are uploaded manually.
The amount of logging that the phone performs can be tuned for the
application to provide more or less detail on specific components of the
phone’s software. For example, if you are troubleshooting a SIP signaling
issue, you are not likely interested in DSP events. Logging levels are adjusted
in the configuration files or via the web interface. You should not modify the
default logging levels unless directed to by Polycom Technical Support.
Inappropriate logging levels can cause performance issues on the phone.
In addition to logging events, the phone can be configured to automatically
execute command-line instructions at specified intervals that output run-time
information such as memory utilization, task status, or network buffer
contents to the log file. These techniques should only be used in consultation
with Polycom Technical Support.
Application Logging Options
Each of the components of the application software is capable of logging
events of different severity. This allows you to capture lower severity events
in one part of the application, while still only getting high severity event for
other components.
The parameters for log level settings are found in the sip.cfg configuration file.
They are log.level.change.module_name. Log levels range from 1 to 6 (1 for
the most detailed logging, 6 for critical errors only).
When testing is complete, remember to return all logging levels to the default
value of 4.
There are other logging parameters that you may wish to modify. Changing
these parameters does not have the same impact as changing the logging
levels, but you should still understand how your changes will affect the phone
and the network.
•
•
log.render.level—Sets the lowest level that can be logged (default=1)
log.render.file.size—Maximum size before log file is uploaded
(default=16 kb)
•
•
log.render.file.upload.period—Frequency of log uploads (default is
172800 seconds = 48 hours)
log.render.file.upload.append—Controls if log files on the boot
server are overwritten or appended, not supported by all servers
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Administrator’s Guide SoundPoint IP / SoundStation IP
•
•
log.render.file.upload.append.sizeLimit—Controls the maximum
size of log files on the boot server (default=512 kb)
log.render.file.upload.append.limitMode—Controls action to take
when server log reaches max size, actions are stop and delete
Scheduled Logging
Scheduled logging is a powerful tool for anyone who is trying to troubleshoot
an issue with the phone that only occurs after some time in operation.
The output of these instructions is written to the application log, and can be
examined later (for trend data).
The parameters for scheduled logging are found in the sip.cfg configuration
file. They are log.sched.module_name
.
The following figure shows an example of a configuration file and the
resulting log file.
Manual Log Upload
If you want to look at the log files without having to wait for the phone to
upload them (which could take as long as 24 hours or more), initiate an upload
by pressing correct combination of keys on the phone.
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Troubleshooting Your SoundPoint IP / SoundStation IP Phones
When the log files are manually uploaded, the word “now” is inserted into the
name of the file, for example, 0004f200360b-now-boot.log .
Reading a Boot Log
The following figure shows a portion of a boot log file:
Boot Failure Messages
The following figure shows an example of “Application sip.ld is not
compatible with this phone!” boot failure messages:
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Troubleshooting Your SoundPoint IP / SoundStation IP Phones
Power and Startup
Symptom
Problem
Corrective Action
There are power issues.
The SoundPoint IP /
Do one of the following:
SoundStation IP family SIP
phone has no power.
•
•
•
Verify that no lights appear on the unit
when it is powered up.
Check if the phone is properly plugged
into a functional AC outlet.
Make sure that the phone isn't
plugged into a plug controlled by a
light switch that is off.
•
•
•
If plugged into a power strip, try
plugging directly into a wall outlet
instead.
Try the phone in another room where
the electricity is known to be working
on a particular outlet.
If using PoE, the power supply voltage
may be too high or too low.
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Administrator’s Guide SoundPoint IP / SoundStation IP
Controls
Symptom
Problem
Corrective Action
The dial pad does not work.
The dial pad on the SoundPoint
IP / SoundStation IP family SIP
phone does not respond.
Do one of the following:
•
Check for a response from other
feature keys or from the dial pad.
•
Place a call to the phone from a known
working telephone. Check for display
updates.
•
•
•
Press the Menu key followed by
System Status and Server Status to
check if the telephone is correctly
registered to the server.
Press the Menu key followed by
System Status and Network Statistics.
Scroll down to see if LAN port shows
active or Inactive.
Check the termination at the switch or
hub end of the network LAN cable.
Ensure that the switch/hub port
connected to the telephone is
operational (if not accessible, contact
your system administrator).
•
Before restarting your phone, contact
your system administrator, since this
may allow more detailed
troubleshooting to occur before losing
any current status information.
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Troubleshooting Your SoundPoint IP / SoundStation IP Phones
Access to Screens and Systems
Symptom
Problem
Corrective Action
There is no response from
feature key presses.
The SoundPoint IP /
SoundStation IP family SIP
phone is not in active state.
Do one of the following:
•
•
Press the keys more slowly.
Check to see whether or not the key
has been mapped to a different
function or disabled.
•
Make a call to the phone to check for
inbound call display and ringing as
normal. If successful, try to press
feature keys within the call to access
Directory or Buddy Status, for
example.
•
•
Press Menu followed by Status >
Lines to confirm line is actively
registered to the call server.
Reboot the phone to attempt re- to the
call server (refer to Rebooting the
The display shows “Network Link
is Down”.
The LAN cable is not properly
connected.
Do one of the following:
•
•
•
Check termination at the switch or hub
(furthest end of the cable from the
phone).
Check that the switch or hub is
operational (flashing link/status lights)
or contact your system administrator.
Press Menu followed by Status >
Network. Scroll down to verify that the
LAN is active.
•
•
Ping phone from another machine.
Reboot the phone to attempt re- to the
call server (refer to Rebooting the
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Administrator’s Guide SoundPoint IP / SoundStation IP
Calling
Symptom
Problem
Corrective Action
There is no dial tone.
Power is not correctly applied to
the SoundPoint IP family SIP
phone.
Do one of the following:
•
•
Check that the display is illuminated.
Make sure the LAN cable is inserted
properly at the rear of the phone (try
unplugging and re-inserting the
cable).
•
If using in-line powering, have your
system administrator check that the
switch is supplying power to the
phone.
Dial tone is not present on one of
audio paths.
Do one of the following:
•
Switch between Handset, Headset (if
present) or Hands-Free
Speakerphone to see if dial tone is
present on another paths.
•
•
If dial tone exists on another path,
connect a different handset or
headset to isolate the problem.
Check configuration for gain levels.
The phone is not registered.
Ring setting or volume is low.
Contact your system administrator.
The phone does not ring.
Do one of the following:
•
•
Adjust the ringing level from the front
panel using the volume up/down keys.
Check same status of handset,
headset (if connected) and through
the Hands-Free Speakerphone.
Outbound or inbound calling is
unsuccessful.
Do one of the following:
•
Place a call to the phone under
investigation. Check that the display
indicates incoming call information.
•
Lift the handset. Ensure dial tone is
present and place a call to another
extension or number. Check that the
display changes in response.
The line icon shows an
unregistered line icon.
The phone line is unregistered.
Contact your system administrator.
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Troubleshooting Your SoundPoint IP / SoundStation IP Phones
Displays
Symptom
Problem
Corrective Action
There is no display.
Power is not correctly applied to
the SoundPoint IP family SIP
phone.
Do one of the following:
The display is incorrect.
The display has bad contrast.
•
•
Check that the display is illuminated.
Make sure the LAN cable is inserted
properly at the rear of the phone (try
unplugging and re-inserting the
cable).
•
If using in-line powering, have your
system administrator check that the
switch is supplying power to the
phone.
The contrast needs adjustment.
Do one of the following:
•
Refer to the appropriate SoundPoint
IP / SoundStation IP SIP phone User
Guide.
•
Reboot the phone to obtain a default
level of contrast (refer to Rebooting
Outbound or inbound calling is
unsuccessful.
Do one of the following:
•
Place a call to the phone under
investigation. Check that the display
indicates incoming call information.
•
Lift the handset. Ensure dial tone is
present and place a call to another
extension or number. Check that the
display changes in response.
The display is flickering.
Certain type of older fluorescent
lighting causes the display to
appear to flicker.
Do one of the following:
•
Move the SoundPoint IP /
SoundStation IP SIP phone away
from the lights.
•
Replace the lights.
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Administrator’s Guide SoundPoint IP / SoundStation IP
Audio
Symptom
Problem
Corrective Action
There is no audio on the
headset.
The connections are not correct.
Do one of the following:
•
•
Ensure the headset is plugged into the
jack marked Headset at the rear of the
phone.
Ensure the headset amplifier (if
present) is turned on and/or the
volume is correctly adjusted).
There are audio and echo issues
on the headset.
Possible issues include:
Refer to “Technical Bulletin 16249:
Troubleshooting Audio and Echo Issues
on SoundPoint® IP Phones” on the
Polycom Support Knowledgebase.
•
•
•
Echo on external calls
through a gateway.
Internal calls (no gateway),
handsfree echo.
Internal calls (no gateway),
handset to handset echo.
Upgrading
Symptom
Problem
Corrective Action
SoundPoint IP 300 and/or 500
behave incorrectly or do not
display new features.
New features are not supported
on the SoundPoint IP 300 and
500 and the configuration files
have not been correctly modified.
The SoundPoint IP 300 and 500
will not ‘understand’ the new
configuration parameters, and
will attempt to load the new
application.
The attempt to load the new application
will fail since there is no 300/500 image
contained within the sip.ld file, so the
phone will continue on and run the current
version of application that it has in
memory. It will however use the new
configuration files. Refer to Supporting
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A
Configuration Files
This appendix provides detailed descriptions of certain configuration files
used by the Session Initiation Protocol (SIP) application. It is a reference for all
parameters that are configurable when using the centralized provisioning
installation model.
This appendix contains information on:
•
•
•
•
The application configuration files dictate the behavior of the phone once it is
running the executable specified in the master configuration file.
Configuration files should only be modified by a knowledgeable system
Caution
administrator. Applying incorrect parameters may render the phone unusable. The
configuration files which accompany a specific release of the SIP software must be
used together with that software. Failure to do this may render the phone unusable.
In the tables in the following sections, “Null” should be interpreted as the empty
string, that is, attributeName=“” when the file is viewed in an XML editor.
Note
To enter special characters in a configuration file, enter the appropriate sequence
using an XML editor:
•
•
•
•
•
& as &
” as "
’ as '
< as <
> as >
The various .hd. parameters in sip.cfg (such as voice.aec.hd.enable
voice.ns.hd.enable, and voice.agc.hd.enable) are headset parameters. They
,
Note
are not connected to high definition or HD voice.
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Administrator’s Guide SoundPoint IP / SoundStation IP
Master Configuration Files
The master configuration files can be one of:
•
Specified master configuration file—The master configuration file can be
explicitly specified in the boot server address, for example,
http://usr:pwd@server/dir/example1.cfg. The filename must end with
.cfg and be at least five characters long. If this file cannot be downloaded,
the phone will search for the per-phone master configuration file
(described next).
•
Per-phone master configuration file—If per-phone customization is
required, the file should be named <Ethernet address>.cfg, where
Ethernet address is the MAC address of the phone in question. For A-F
hexadecimal digits, use upper or lower case, for example,
0004f200106c.cfg. The Ethernet address can be viewed using the About
soft key during the auto-restart countdown of the bootROM or through
the Menu > Status > Platform > Phone menu in the application. It is also
printed on a label on the back of the phone. If this file cannot be
downloaded, the phone will search for the default master configuration
file (described next).
•
Default master configuration file—For systems in which the configuration
is identical for all phones (no per-phone <Ethernet address>.cfg files), the
default master configuration file may be used to set the configuration for
all phones. The file named 000000000000.cfg (<12 zeros>.cfg) is the default
master configuration file and it is recommended that one be present on the
boot server. If a phone does not find its own <Ethernet address>.cfg file,
it will use this one, and establish a baseline configuration. This file is part
of the standard Polycom distribution of configuration files. It should be
used as the template for the <Ethernet address>.cfg files.
The default master configuration file, 000000000000.cfg, is shown below:
<?xml version=”1.0” standalone=”yes”?>
<!-- Default Master SIP Configuration File -->
<!-- edit and rename this file to <Ethernet-address>.cfg for
each phone. -->
<!-- $Revision: 1.14 $ $Date 2005/07/27 18:43:30 $ -->
< APPLICATION APP_FILE_PATH=”sip.ld”
CONFIG_FILES=”phone1.cfg, sip.cfg” MISC_FILES=””
LOG FILE DIRECTORY=”” OVERRIDES_DIRECTORY=””
CONTACTS_DIRECTORY=”” LICENSE_DIRECTORY=””/>
Master configuration files contain six XML attributes:
•
APP_FILE_PATH—The path name of the application executable. It can
have a maximum length of 255 characters. This can be a URL with its own
protocol, user name and password, for example
http://usr:pwd@server/dir/sip.ld.
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Configuration Files
•
•
CONFIG_FILES—A comma-separated list of configuration files. Each file
name has a maximum length of 255 characters and the list of file names has
a maximum length of 2047 characters, including commas and white space.
Each configuration file can be specified as a URL with its own protocol,
user name and password, for example
ftp://usr:pwd@server/dir/phone2034.cfg.
MISC_FILES—A comma-separated list of other required files. Dictionary
resource files listed here will be stored in the phone's flash file system. So
if the phone reboots at a time when the boot server is unavailable, it will
still be able to load the preferred language.
On the SoundPoint IP 500, there is insufficient room for a language file. Specifying
one will cause a reboot loop
Note
•
•
LOG_FILE_DIRECTORY—An alternative directory to use for log files if
required. A URL can also be specified. This is blank by default.
CONTACTS_DIRECTORY—An alternative directory to use for user
directory files if required. A URL can also be specified. This is blank by
default.
•
•
OVERRIDES_DIRECTORY—An alternative directory to use for
configuration overrides files if required. A URL can also be specified. This
is blank by default.
LICENSE_DIRECTORY—An alternative directory to use for license files if
required. A URL can also be specified. This is blank by default.
The order of the configuration files listed in CONFIG_FILES is significant:
Warning
•
•
•
The files are processed in the order listed (left to right).
The same parameters may be included in more than one file.
The parameter found first in the list of files will be the one that is effective.
This provides a convenient means of overriding the behavior of one or more phones
without changing the baseline configuration files for an entire system.
IP Phones” whitepaper at www.polycom.com/support/voice/
If you have a requirement for different application loads on different phones
on the same boot server, you can create a variable in the master configuration
file that is replaced by the MAC address of each phone when it reboots. An
example is shown below:
<?xml version=”1.0” standalone=”yes”?>
<!-- Default Master SIP Configuration File -->
<!-- edit and rename this file to <Ethernet-address>.cfg for
each phone. -->
<!-- $RCSfile: 000000000000.cfg,v $ $Revision:$ -->
< APPLICATION APP_FILE_PATH=”sip[MACADDRESS].ld”
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Administrator’s Guide SoundPoint IP / SoundStation IP
CONFIG_FILES=”phone1[MACADDRESS].cfg, sip.cfg” MISC_FILES=””
LOG FILE DIRECTORY=”” OVERRIDES_DIRECTORY=””
CONTACTS_DIRECTORY=”” LICENSE_DIRECTORY=””/>
If you have a requirement for separate application loads on different phones
on the same boot server, you can modify the application that is loaded when
each phone reboots. An example is below:
<?xml version=”1.0” standalone=”yes”?>
<!-- Default Master SIP Configuration File -->
<!-- edit and rename this file to <Ethernet-address>.cfg for
each phone. -->
<!-- $RCSfile: 000000000000.cfg,v $ $Revision:$ -->
< APPLICATION APP_FILE_PATH=”sip[PHONE_PART_NUMBER].ld”
CONFIG_FILES=”phone1.cfg, sip.cfg” MISC_FILES=””
LOG FILE DIRECTORY=”” OVERRIDES_DIRECTORY=””
CONTACTS_DIRECTORY=”” LICENSE_DIRECTORY=””/>
You can also use the substitution strings PHONE_MODEL,
PHONE_PART_NUMBER, and PHONE_MAC_ADDRESS in the master
configuration file.
You can also direct phone upgrades to a software image and configuration
files based on the phone model number and part number. All XML attributes
can be modified in this manner. An example is below:
<?xml version=”1.0” standalone=”yes”?>
<!-- Default Master SIP Configuration File -->
<!-- edit and rename this file to <Ethernet-address>.cfg for
each phone. -->
<!-- $RCSfile: 000000000000.cfg,v $ $Revision:$ -->
<APPLICATION APP_FILE_PATH=”sip.ld” CONFIG_FILES=”phone1.cfg,
sip.cfg” MISC_FILES=”” LOG_FILE_DIRECTORY=””
OVERRIDES_DIRECTORY=””
CONTACTS_DIRECTORY=”” LICENSE_DIRECTORY=””
APP_FILE_PATH_SPIP330=”SPIP330.sip.ld”
CONFIG_FILES_SPIP330=”phone1_SPIP330.cfg, sip_SPIP330.cfg”
APP_FILE_PATH_SPIP501=”SPIP501.sip.ld”
CONFIG_FILES_SPIP501=”phone1_SPIP501.cfg, sip_SPIP501.cfg” />
For more information, refer to “Technical Bulletin 35361: Overriding
Parameters in Master Configuration File on SoundPoint IP Phones“ at
Application Configuration
The configuration file sip.cfg contains SIP protocol and core configuration
settings that would typically apply to an entire installation and must be set
before the phones will be operational, unless changed through the local web
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Configuration Files
server interface or local menu settings on the phone. These settings include the
local port used for SIP signaling, the address and ports of a cluster of SIP
application servers, voice codecs, gains, and tones, and other parameters.
These parameters include:
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
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Administrator’s Guide SoundPoint IP / SoundStation IP
•
•
Polycom recommends that you create another file with your organization’s
modifications. If you must change any Polycom templates, back them up first.
Protocol <volpProt/>
This attribute includes:
•
•
•
•
Local <local/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
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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Configuration Files
Server <server/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
voIpProt.server.dhcp.available
0, 1
0
If set to 1, check with the DHCP server for
SIP server IP address. If set to 0, do not
check with DHCP server.
voIpProt.server.dhcp.option
128 to 255
Null
Option to request from the DHCP server if
voIpProt.server.dhcp.available = 1. There is
no default value for this parameter, it must be
filled in with a valid value.
Note: If the reg.x.server.y.address parameter
non-Null, it takes precedence even if the
DHCP server is available.
voIpProt.server.dhcp.type
0, 1
Null
If set to 0, IP request address.
If set to 1, request string.
Type to request from the DHCP server if
voIpProt.server.dhcp.available = 1.
There is no default value for this parameter, it
must be filled in with a valid value.
voIpProt.server.x.address
voIpProt.server.x.port
dotted-deci
mal IP
address or
host name
Null
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.
Note: If the reg.x.server.y.address parameter
non-Null, all of the reg.x.server.y.xxx
parameters will override the voIpProt.server
parameters.
0, Null, 1 to
65535
If port is 0 or Null:
If voIpProt.server.x.addressis a
hostname and
voIpProt.server.x.transport is set to
DNSnaptr, do NAPTR then SRV lookups.
If voIpProt.server.x.transportis set to
TCPpreferred or UDPOnly then use 5060
and don’t advertise the port number in
signalling.
If voIpProt.server.x.addressis an IP
address, there is no DNS lookup and 5060 is
used for the port but it is not advertised in
signaling.
If port is 1 to 65535:
This value is used and it is advertised in
signaling.
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Permitted
Attribute
Values
Default
Interpretation
voIpProt.server.x.transport
DNSnaptr or
TCPpreferre
d or
UDPOnly or
TLS or
DNSnapt If set to Null or DNSnaptr:
r
If voIpProt.server.x.addressis a
hostname and voIpProt.server.x.port is 0 or
Null, do NAPTR then SRV look-ups to try to
discover the transport, ports and servers, as
per RFC 3263. If
TCPOnly
voIpProt.server.x.addressis an IP
address, or a port is given, then UDP is used.
If set to TCPpreferred:
TCP is the preferred transport, UDP is used if
TCP fails.
If set to UDPOnly:
Only UDP will be used.
If set to TLS:
If TLS fails, transport fails. Leave port field
empty (will default to 5061) or set to 5061.
If set to TCPOnly:
Only TCP will be used.
NOTE: TLS is not supported on SoundPoint
IP 300 and 500 phones.
voIpProt.server.x.expires
positive
integer,
minimum
300
3600
The phone’s requested registration period in
seconds.
Note: The period negotiated with the server
may be different. The phone will attempt to
re-register at the beginning of the overlap
period. For example, if “expires”=3600 and
“overlap”=60, the phone will re-register after
3540 seconds (3600 – 60).
voIpProt.server.x.expires.overlap
positive
integer,
minimum 5,
maximum
65535
60
The number of seconds before the expiration
time returned by server x at which the phone
should try to re-register. The phone will try to
re-register at half the expiration time returned
by the server if that value is less than the
configured overlap value.
voIpProt.server.x.register
0, 1
1
0
If set to 0, calls can be routed to an outbound
proxy without registration.
voIpProt.server.x.retryTimeOut
Null or
non-negativ
e integer
If set to 0 or Null, use standard RFC 3261
signaling retry behavior. Otherwise
retryTimeOut determines how often retries
will be sent.
Units = milliSeconds. (Finest resolution =
100ms).
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Configuration Files
Permitted
Values
Attribute
Default
Interpretation
voIpProt.server.x.retryMaxCount
Null or
non-negativ
e integer
3
If set to 0 or Null, 3 is used. retryMaxCount
retries will be attempted before moving on to
the next available server.
voIpProt.server.x.expires.lineSeize
voIpProt.server.x.lcs
positive
integer,
minimum 10
30
0
Requested line-seize subscription period.
0, 1
This attribute overrides the
voIpProt.SIP.lcs
.
If set to 1, the proprietary “epid” parameter is
added to the From field of all requests to
support Microsoft Live Communications
Server.
SDP <SDP/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
volpProt.SDP.answer.userLocalPrefe
rences
0 or 1
0
If set to 1, the phones uses its own
preference list when deciding which codec to
use rather than the preference list in the offer.
If set to 0, it is disabled.
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SIP <SIP/>
This configuration attribute is defined as follows:
Permitted
Values
Attribute
Default
Interpretation
voIpProt.SIP.useContactInReferTo
0, 1
0
If set to 0, the “To URI” is used in the REFER.
If set to 1, the “Contact URI” is used in the
REFER.
voIpProt.SIP.useRFC2543hold
voIpProt.SIP.useSendonlyHold
0, 1
0
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 initiating
hold. In either case, the phone processes
incoming hold signaling in either format.
0, 1
If set to 1, the phone will send a reinvite with
a stream mode attribute of “sendonly” when a
call is put on hold. This is the same as the
previous behavior.
If set to 0, the phone will send a reinvite with
a stream mode attribute of “inactive” when a
call is put on hold.
NOTE: The phone will ignore the value of this
parameter if set to 1 when the parameter
voIpProt.SIP.useRFC2543holdis also set
to 1 (default is 0).
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 Microsoft Live Communications
Server.
voIpProt.SIP.ms-forking
If set to 0, support for MS-forking is disabled.
If set to 1, support for MS-forking is enabled
and the phone will reject all Instant Message
INVITEs. This parameter is relevant for
Microsoft Live Communications Server
server installations.
Note that if any end point registered to the
same account has MS-forking disabled, all
other end points default back to non-forking
mode. Windows Messenger does not use
MS-forking so be aware of this behavior if
one of the end points is Windows Messenger.
voIpProt.SIP.dialog.usePvalue
0, 1
0
If set to 0, phone uses "pval" field name in
Dialog. This obeys the
draft-ietf-sipping-dialog-package-06.txt draft.
If set to 1, phone uses a field name of
"pvalue".
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Configuration Files
Permitted
Values
Attribute
Default
Interpretation
voIpProt.SIP.connectionReuse.useAli
as
0, 1
0
If set to 0, this is the old behavior.
If set to 1, phone uses the connection reuse
draft which introduces "alias".
voIpProt.SIP.sendCompactHdrs
0, 1
0
0
If set to 0, SIP header names generated by
the phone use the long form, for example
‘From’.
If set to 1, SIP header names generated by
the phone use the short form, for example ‘f’.
voIpProt.SIP.keepalive.sessionTimer
s
0, 1
If set to 1, the session timer will be enabled.
If set to 0, the session timer will be disabled,
and the phone will not declare “timer” in
“Support” header in INVITE. The phone will
still respond to a re-INVITE or UPDATE. The
phone will not try to re-INVITE or do UPDATE
even if remote end point asks for it.
voIpProt.SIP.requestURI.E164.addGl
obalPrefix
0, 1
0, 1
0
1
If set to 1, ‘+’ global prefix is added to E.164
user parts in sip: URIs:.
voIpProt.SIP.allowTransferOnProcee
ding
If set to 1, a transfer can be completed during
the proceeding state of a consultation call.
If set to 0, a transfer is not allowed during the
proceeding state of a consultation call.
If set to Null, the default value is used.
voIpProt.SIP.dialog.useSDP
0, 1
0
If set to 0, new dialog event package draft is
used (no SDP in dialog body).
If set to 1, for backwards compatibility, use
this setting to send SDP in dialog body.
voIpProt.SIP.pingInterval
0 to 3600
0, 1
0
0
The number in seconds to send "PING"
message. This feature is disabled by default.
voIpProt.SIP.useContactInReferTo
If set to 1, the Contact URI is used.
If set to 0, the TO URI is used (previous
behavior).
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Permitted
Attribute
Values
Default
Interpretation
voIpProt.SIP.serverFeatureControl.cf
0, 1
0
If set to 1, server-based call forwarding is
enabled. The call server has control of call
forwarding.
If set to 0, server-based call forwarding is not
enabled. This is the old behavior.
voIpProt.SIP.serverFeatureControl.dn 0, 1
d
0
0
If set to 1, server-based DND is enabled. The
call server has control of DND.
If set to 0, server-based DND is not enabled.
This is the old behavior.
voIpProt.SIP.authOptimizedInFailover 0,1
If set to 1, when failover occurs, the first new
SIP request is sent to the server that sent the
proxy authentication request.
If set to 0, when failover occurs, the first new
SIP request is sent to the server with the
highest priority in the server list.
If reg.x.auth.optimizedInFailoverset to
Null, this attribute is checked.
If
voIpProt.SIP.authOptimizedInFailover
is Null, then this feature is disabled.
If both attributes are set, the value of
reg.x.auth.optimizedInFailovertakes
precedence.
This attribute also includes:
•
•
•
•
•
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Configuration Files
Outbound Proxy <outboundProxy/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
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
voIpProt.SIP.outboundProxy.transpor
t
DNSnaptr or
TCPpreferre
d or
DNSnapt If set to Null or DNSnaptr:
If voIpProt.SIP.outboundProxy.address is a
r
hostname and
UDPOnly or
TLS or
TCPOnly
voIpProt.SIP.outboundProxy.portis 0 or
Null, do NAPTR then SRV look-ups to try to
discover the transport, ports and servers, as
per RFC 3263. If
voIpProt.SIP.outboundProxy.addressis
an IP address, or a port is given, then UDP is
used.
If set to TCPpreferred:
TCP is the preferred transport, UDP is used if
TCP fails.
If set to UDPOnly:
Only UDP will be used.
If set to TLS:
If TLS fails, transport fails. Leave port field
empty (will default to 5061) or set to 5061.
If set to TCPOnly:
Only TCP will be used.
NOTE: TLS is not supported on SoundPoint
IP 300 and 500 phones.
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Alert Information <alertInfo/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
volpProt.SIP.alertInfo.x.value
string to
Null
Alert-Info fields from INVITE requests will be
compared against as many of these
parameters as are specified (x=1, 2, ..., N)
and if a match is found, the behavior
described in the corresponding ring class
be applied.
compare
against the
value of
Alert-Info
headers in
INVITE
requests
voIpProt.SIP.alertInfo.x.class
positive
integer
Null
Request Validation <requestValidation/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
voIpProt.SIP.requestValidation.x.req One of:
uest “INVITE”, “ACK”
Null
Sets the name of the method for which
validation will be applied.
, “BYE”
WARNING: Intensive request validation
may have a negative performance impact
due to the additional signaling required in
some cases, therefore, use it wisely.
“REGISTER”,
“CANCEL”,
“OPTIONS”,
“INFO”,
“MESSAGE”,
“SUBSCRIBE”
“NOTIFY”,
“REFER”,
“PRACK”, or
“UPDATE”
voIpProt.SIP.requestValidation.x.me Null or
Null
If Null, no validation is done. Otherwise this
sets the type of validation performed for the
request:
thod
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 servers;
digest: challenge requests with digest
authentication using the local credentials
for the associated registration (line);
both or all: apply both of the above methods
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Configuration Files
Permitted
Values
Attribute
Default
Interpretation
voIpProt.SIP.requestValidation.x.req A valid string
uest.y.event
Null
Determines which events specified with the
Event header should be validated; only
applicable when
voIpProt.SIP.requestValidation.x.re
questis set to “SUBSCRIBE” or “NOTIFY”.
If set to Null, all events will be validated.
voIpProt.SIP.requestValidation.dige
st.realm
A valid string
Polycom Determines string used for Realm.
SPIP
Special Events <specialEvent/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
voIpProt.SIP.specialEvent.lineSeize.n 0, 1
onStandard
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 processing.
voIpProt.SIP.specialEvent.checkSync 0, 1
.alwaysReboot
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.
Conference Setup <conference/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
voIpProt.SIP.conference.address
ASCII string
up to 128
characters
long
Null
If Null, conferences are set up on the phone
locally.
If set to some value, conferences are set up
by the server using the conferencing agent
specified by this address. The acceptable
values depend on the conferencing server
implementation policy.
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Dial Plan <dialplan/>
The dial plan is not applied against Placed Call List, VoiceMail, last call return, and
remote control dialed numbers.
Note
This configuration attribute is defined as follows:
Permitted
Attribute
dialplan.applyToCallListDial
Values
Default
Interpretation
0, 1
0
This attribute covers dialing from Received
Call List and Missed Call List including dialing
from Edit or Info sub- menus.
If set to 0, the dial plan is not applied against
the dialed number.
if set to 1, the dial plan is applied against the
dialed number.
dialplan.applyToDirectoryDial
0, 1
0
This attribute covers dialing from Directory as
well as Speed Dial List.
Value interpretation is the same as for
dialplan.applyToCallListDial
.
Note: An Auto Call Contact number is
considered a dial from directory.
dialplan.applyToUserDial
0, 1
1
1
0
This attribute covers the case when the user
presses the Dial soft key to send dialed
number when in idle state display.
Value interpretation is the same as for
dialplan.applyToCallListDial.
dialplan.applyToUserSend
dialplan.impossibleMatchHandling
0, 1
This attribute covers the case when the user
presses the Send soft key to send the dialed
number.
Value interpretation is the same as for
dialplan.applyToCallListDial.
0, 1 or 2
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.
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Configuration Files
This attributes also includes:
•
•
Digit Map <digitmap/>
A digit map is defined either by a “string” or by a list of strings. Each string in
the list is an alternative numbering scheme, specified either as a set of digits or
timers, or as an expression over which the gateway will attempt to find a
shortest possible match.
Digit map extension letter “R” indicates that certain matched strings are
replaced. The following examples shows the semantics of the syntax:
•
•
R9RRxxxxxxx—Remove 9 at the beginning of the dialed number
— For example, if a customer dials 914539400, the first 9 is removed
when the call is placed.
RR604Rxxxxxxx—Prepend 604 to all 7 digit numbers
— For example, if a customer dials 4539400, 604 is added to the front of
the number, so a call 6044539400 is placed.
•
•
•
•
R9R604Rxxxxxxx—Replaces 9 with 604
R999R911R—Convert 999 to 911
xxR601R600Rxx—When applied on 1160122 gives 1160022
xR60xR600Rxxxxxxx—Any 60x will be replaced with 600 in the middle of
the dialed number that matches
— For example, if a customer dials 16092345678, a call is placed to
16002345678.
•
911xxx.T— A period (".") which matches an arbitrary number, including
zero, of occurrences of the preceding construct
— For example:
91112 with waiting time to comply with T is a match
911123 with waiting time to comply with T is a match
9111234 with waiting time to comply with T is a match
and the number can grow indefinitely given that pressing the next
digit takes less than T.
The following guidelines should be noted:
•
•
You must use only *, #, or 0-9 between second and third R
If a digit map does not comply, it is not included in the digit plan as a valid
one. That is, no matching is done against it.
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•
•
There is no limitation on the number of R triplet sets in a digit map.
However, a digit map that contains less than full number of triplet sets (for
example, a total of 2Rs or 5Rs) is considered an invalid digit map.
Using T in the left part of RRR syntax is not recommended. For example,
R0TR322R should be avoided.
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
dialplan.digitmap
string compatible with the
digit map feature of
[2-9]11|0T|
When this attribute is
present, number-only dialing
during the setup phase of
new calls will be compared
against the patterns therein
and if a match is found, the
call will be initiated
+011xxx.T|
MGCP described in 2.1.5
of RFC 3435. String is
limited to 768 bytes and
30 segments; a comma is
also allowed; when
reached in the digit map,
a comma will turn dial
tone back on;’+’ is allowed
as a valid digit; extension
letter ‘R’ is used as
0[2-9]xxxxxxxxx|
+1[2-9]xxxxxxxx|
[2-9]xxxxxxxxx|
[2-9]xxxT
automatically eliminating the
need to press Send.
Attributes
dialplan.applyToCallLis
tDial
dialplan.applyToDirecto
ryDial
dialplan.applyToUserDia
, and
dialplan.applyToUserSen
control the use of match
,
defined above.
,
l
d
and replace in the dialed
number in the different
scenarios.
dialplan.digitmap.timeOut
string of positive integers
separated by ‘|’
3 | 3 | 3 | 3 | 3 | 3
Timeout in seconds for each
segment of digit map.
Note: If there are more digit
maps than timeout values,
the default value of 3 will be
used. If there are more
timeout values than digit
maps, the extra timeout
values are ignored.
Routing <routing/>
This attribute allows the user to create a specific routing path for outgoing SIP
calls independent of other “default” configurations.
This attribute also includes:
•
•
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Configuration Files
Server <server/>
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
dialplan.routing.server.x.addre dotted-decimal IP address
Null
IP address or host name and port of
a SIP server that will be used for
routing calls. Multiple servers can
be listed starting with x=1, 2, ... for
fault tolerance.
ss
or host name
1 to 65535
dialplan.routing.server.x.port
5060
Emergency <emergency/>
In the following attributes, x is the index of the emergency entry description
and y is the index of the server associated with emergency entry x. For each
emergency entry (index x), one or more server entries (indexes (x,y)) can be
configured. x and y must both use sequential numbering starting at 1.
Attribute
Permitted Values
Default
Interpretation
dialplan.routing.emergency.x.
value
Single entry representing
a SIP URL
for x =1,
value = “911”, Null
for all others
This determines the URLs
that should be watched for.
When one of these defined
URLs is detected as having
been dialed by the user, the
call will automatically be
directed to the defined
emergency server.
dialplan.routing.emergency.x.
server.y
positive integer
for x=1, y =1, Null
for all others
Index representing the
will be used for emergency
routing.
Localization <lcl/>
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
This attribute includes:
•
•
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Multilingual <ml/>
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.
Attribute
Permitted Values
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 specified in the
format language-region.
An exact match for
one of the folder
names under the
SoundPointIPLocalizat
ion folder on the boot
server.
lcl.ml.lang.menu.x
String in the format
language_region
Multiple lcl.ml.lang.menu.x attributes
are supported - as many languages as are
desired. However, the lcl.ml.lang.menu.x
attributes must be sequential
(
lcl.ml.lang.menu.1
,
lcl.ml.lang.menu.2
,
lcl.ml.lang.menu.3, ...,
lcl.ml.lang.menu.N) with no gaps and the
strings must exactly match a folder name
under the SoundPointIPLocalization folder
on the boot server for the phone to be able to
locate the dictionary file.
lcl.ml.lang.clock.x.24HourClock
lcl.ml.lang.clock.x.format
0,1
If attribute present, overrides
lcl.datetime.time.24HourClock.
If 1, display time in 24-hour clock mode
rather than am/pm.
string which includes
‘D’, ‘d’ and ‘M’ and two
optional commas
If attribute present, overrides
lcl.datetime.date.format;
D = day of week
d = day
M = month
Up to two commas may be included.
For example: D,dM = Thursday, 3 July or
Md,D = July 3, Thursday
The field may contain 0, 1 or 2 commas
which can occur only between characters
and only one at a time. For example: “D,,dM”
is illegal.
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Configuration Files
Attribute
Permitted Values
Interpretation
lcl.ml.lang.clock.x.longFormat
0, 1
If attribute present, overrides
lcl.datetime.date.longFormat
.
If 1, display the day and month in long format
(Friday/November), otherwise use
abbreviations (Fri/Nov).
lcl.ml.lang.clock.x.dateTop
lcl.ml.lang.y.list
0, 1
If attribute present, overrides
lcl.datetime.date.dateTop
.
If 1, display date above time, otherwise
display time above date.
“All” or a
comma-separated list
A list of the languages supported on
hardware platform ‘y’ where ‘y’ can be
IP_500 or IP_600.
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 the tables below.
3. Place the file in an appropriately named folder according to the format
language_region parallel to the other dictionary files under the
SoundPointIPLocalization folder on the boot server.
4. Add a lcl.ml.lang.clock.menu.xattribute 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.dateTopattributes and set them according
to the regional preferences.
6. (Optional) Set lcl.ml.langto be the new language_region string.
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
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Extended character support available on SoundPoint IP 600 and SoundStation IP 4000 and 7000 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
Within a Unicode range, some characters may not be supported due to their
infrequent usage
Note
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Configuration Files
Date and Time <datetime/>
This configuration attribute is defined as follows:
Permitted
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.
D = day of week
d = day
includes ‘D’, ‘d’
and ‘M’ and two
optional commas
M = month
Up to two commas may be included.
For example: D,dM = Thursday, 3 July or Md,D = July
3, Thursday
The field may contain 0, 1 or 2 commas which can
occur only between characters and only one at a time.
For example: “D,,dM” is illegal.
lcl.datetime.date.longFormat
lcl.datetime.date.dateTop
0,1
If 1, display the day and month in long format
(Friday/November), otherwise, use abbreviations
(Fri/Nov).
0, 1
If 1, display date above time else display time above
date.
User Preferences <up/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
up.headsetMode
0,1
0
If set to 1, the headset will be selected as the
preferred transducer after its first use until the
headset key is pressed again; otherwise,
hands-free will be selected preferentially over
the headset.
up.useDirectoryNames
0,1
0
If set to 1, the name fields of directory entries
which match incoming calls will be used for
caller identification display and in the call lists
instead of the name provided through network
signaling.
up.oneTouchVoiceMail
0, 1
0, 1
0
1
If set to 1, the voice mail summary display is
bypassed and voice mail is dialed directly (if
configured).
up.welcomeSoundEnabled
If set to 1, play welcome sound effect after a
reboot.
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Permitted
Values
Attribute
Default
Interpretation
up.welcomeSoundOnWarmBootE
nabled
0, 1
0
If set to 1, play welcome sound effect on warm
as well as cold boots, otherwise only a cold
boot will trigger the welcome sound effect.
up.localClockEnabled
up.backlight.onIntensity
0, 1
1
If set to 1, display the date and time on the idle
display
0 (off),
1 (low),
2 (medium), 3
(high)
Null
This parameter controls the intensity of the
LCD backlight when it turns on during normal
use of the phone.
The default value is medium.
up.backlight.idleIntensity
0 (off),
1 (low),
2 (medium), 3
(high)
Null
Null
This parameter controls the intensity of the
LCD backlight when the phone is idle.
The default value is low.
Note: If idleIntensity is set higher than
onIntensity, it will be replaced with the
onIntensity value.
up.idleTimeout
positive
integer,
seconds
Timeout for the idle display or default call
handling display.
If set to 0, there is no timeout.
If set to Null, the default timeout of 20 seconds
is used.
If set to value greater than 0, the timeout is for
that number of seconds (maximum 65536).
up.mwiVisible
0 - Disabled
1 - Enabled
0
If set to 0 or Null, there is no MWI for
registration x (SIP 2.1.0 and 2.1.1 behavior).
If set to 1, msg.mwi.x.callBackModeis set to
disabled. MWI notification will be displayed for
registration x (Pre-SIP 2.1.0 behavior).
Tones <tones/>
This attribute describes configuration items for the tone resources available in
the phone.
This attribute includes:
•
•
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Configuration Files
Dual Tone Multi-Frequency <DTMF/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
tone.dtmf.level
-33 to -3
-15
Level of the high frequency component 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 substituted
with a non-DTMF pacifier tone when
dialing in hands-free mode. This prevents
DTMF digits being broadcast to other
surrounding telephony devices or being
inadvertently transmitted in-band due to
local acoustic echo.
Note: tone.dtmf.chassis.masking should
only be enabled when tone.dtmf.viaRtp is
disabled.
tone.dtmf.stim.pac.offHookOnly
0, 1
0
Not currently used.
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Permitted
Attribute
Values
Default
Interpretation
tone.dtmf.viaRtp
0, 1
1
If set to 1, encode DTMF in the active
RTP stream, otherwise, DTMF may be
encoded within the signaling protocol only
when the protocol offers the option.
Note: tone.dtmf.chassis.masking should
be enabled when tone.dtmf.viaRtp is
disabled.
tone.dtmf.rfc2833Control
0, 1
1
If set to 1, the phone will indicate a
preference for encoding DTMF through
RFC 2833 format in its 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.
Chord-Sets <chord/>
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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Configuration Files
In the following table, x is the chord-set number and cat is one of callProg,
ringer, or misc.
Permitted
Attribute
Values
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.
Sampled Audio for Sound Effects <saf/>
The following sampled audio WAVE file (.wav) formats are supported:
•
•
•
mono 8 kHz G.711 μ-Law
G.711 A-Law
L16/16000 (16-bit, 16 kHz sampling rate, mono)
L16/16000 is not supported on SoundPoint IP 301phones, and SoundStation IP
4000 phones.
Note
The phone uses built-in wave files for some sound effects. The built-in wave
files can be replaced with files downloaded from the boot server or from the
Internet, however, these are stored in volatile memory so the files will need to
remain accessible should the phone need to be rebooted. Files will be
truncated to a maximum size of 300 kilobytes.
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In the following table, x is the sampled audio file number.
Attribute
Permitted Values
Interpretation
saf.x
Null OR valid path name
OR an RFC
1738-compliant URL to a
HTTP, FTP, or TFTP
wave file resource.
If Null, the phone will use a built-in file.
If set to a path name, the phone will attempt to download this file
at boot time from the boot server.
If set to a URL, the phone will attempt to download this file at boot
time from the Internet.
Note: Refer to the above
wave file format
restrictions.
Note: A TFTP URL is expected to be in the format:
tftp://<host>/[pathname]<filename>, for example:
tftp://somehost.example.com/sounds/example.wav .
The following table defines the default usage of the sampled audio files with
the phone:
Sampled Audio File
Default use within phone (pattern reference)
Welcome Sound Effect (se.pat.misc.7)
Ringer 13 (se.pat.ringer.13)
Ringer 14 (se.pat.ringer.14)
Ringer 15 (se.pat.ringer.15)
Ringer 16 (se.pat.ringer.16)
Ringer 17 (se.pat.ringer.17)
Ringer 18 (se.pat.ringer.18)
Ringer 19 (se.pat.ringer.19)
Ringer 20 (se.pat.ringer.20)
Ringer 21 (se.pat.ringer.21)
Ringer 22 (se.pat.ringer.22)
Not used.
1
2
3
4
5
6
7
8
9
10
11
12-24
Sound Effects <se/>
The phone uses both synthesized (based on the chord-sets, refer to Chord-Sets
defined by patterns: rudimentary sequences of chord-sets, silence periods, and
wave files.
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Configuration Files
This configuration attribute is defined as follows:
Permitted
Attribute
Values
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.
This attribute also includes:
•
•
Patterns <pat/>
Patterns use a simple script language that allows different chord sets or wave
files to be strung together with periods of silence. The script language uses the
following instructions:
Instruction
Meaning
Example
sampled (n)
Play sampled audio file
n
se.pat.callProg.x.inst.y.type=”sampled” (sampled audio
file instruction type)
se.pat.callProg.x.inst.y.value =”3” (specifies sampled
audio file 3)
chord (n, d)
Play chord set n (d is
optional and allows the
chord set ON duration to
be overridden to d
milliseconds)
se.pat.callProg.x.inst.y.type= “chord” (chord set
instruction type)
se.pat.callProg.x.inst.y.value= “3” (specifies call
progress chord set 3)
se.pat.callProg.x.inst.y.param= “2000” (override ON
duration of chord set to 2000 milliseconds)
silence (d)
branch (n)
Play silence for d
milliseconds (Rx audio
is not muted)
se.pat.callProg.x.inst.y.type= “silence” (silence
instruction type)
se.pat.callProg.x.inst.y.value= “300” (specifies silence is
to last 300 milliseconds)
Advance n instructions
and execute that
se.pat.callProg.x.inst.y.type= “branch” (branch
instruction type)
instruction (n must be
negative and must not
branch beyond the first
instruction)
se.pat.callProg.x.inst.y.value= “-5” (step back 5
instructions and execute that instruction)
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Currently, patterns that use the sampled instruction are limited to the following
format: sampled followed by optional silence and optional branch back to the
Note
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
Permitted
Attribute
Values
Interpretation
se.pat.callProg.x.name
UTF-8
encoded
string
Used for identification purposes in the user interface (currently
used for ringer patterns only); for patterns that use a sampled
audio file which has been overridden by a downloaded
replacement, the se.pat.ringer.x.name parameter will be
overridden in the user interface by the file names of the wave file.
se.pat.callProg.x.inst.y.type
sampled OR
chord OR
silence OR
branch
As above.
se.pat.callProg.x.inst.y.valu
e
integer
Instruction type:
sampled
chord
Interpretation:
sampled audio file number
chord set number
silence
silence duration in ms
number of instructions to advance
branch
se.pat.callProg.x.inst.y.para
m
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.
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
3
4
5
6
busy tone
ring back tone
reorder tone
stuttered dial tone
call waiting tone
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Configuration Files
Call progress
pattern number
Use within phone
7
alternate call waiting tone (distinctive)
confirmation tone
8
9
howler tone (off-hook warning)
record warning
10
11
12
13
14
15
message waiting tone
alerting
intercom announcement tone
barge-in tone
secondary dial tone
Ringer Patterns
The following table maps ringer pattern numbers to their default descriptions.
Ringer pattern number
Default description
Silent Ring
1
2
Low Trill
3
Low Double Trill
Medium Trill
4
5
Medium Double Trill
High Trill
6
7
High Double Trill
Highest Trill
8
9
Highest Double Trill
Beeble
10
11
12
13
14
15
16
17
Triplet
Ringback-style
Sampled audio file 2
Sampled audio file 3
Sampled audio file 4
Sampled audio file 5
Sampled audio file 6
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Ringer pattern number
Default description
Sampled audio file 7
Sampled audio file 8
Sampled audio file 9
Sampled audio file 10
Sampled audio file 11
18
19
20
21
22
Silent Ring will only provide a visual indication of an incoming call, but no audio
indication.
Note
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, refer to Sampled Audio for
Miscellaneous Patterns
The following table maps miscellaneous patterns to their usage within the
phone.
Miscellaneous
pattern number
Use within phone
new message waiting indication
new instant message
Not used
1
2
3
4
5
6
7
local hold notification
positive confirmation
negative confirmation
welcome (boot up)
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Configuration Files
Ring type <rt/>
Ring type is used to define a simple class of ring to be applied based on some
credentials that are usually carried within the network protocol. The ring class
includes attributes such as call-waiting and ringer index, if appropriate. The
ring class can use one of four types of ring that are defined as follows:
ring
Play a specified ring pattern or call waiting indication.
visual
Provide only a visual indication (no audio indication) of incoming call (no
ringer needs to be specified).
answer
Provide auto-answer on incoming call.
ring-answer
Provide auto answer on incoming call after a ring period.
The auto-answer on incoming call is currently only applied if there is no other call in
progress on the phone at the time.
Note
In the following table, x is the ring class number. The x index needs to be
sequential.
Attribute
Permitted Values
Interpretation
se.rt.enabled
0,1
Set to 1 to enable the ring type feature within the
phone, 0 otherwise.
se.rt.modification.enabled
0,1
Set to 1 to allow user modification through local
user interface of the pre-defined ring type enabled
for modification.
se.rt.x.name
se.rt.x.type
se.rt.x.ringer
UTF-8 encoded string
Used for identification purposes in the user
interface.
ring OR visual OR answer
OR ring-answer
As defined in table above.
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 Ringer
se.rt.x.callWait
se.rt.x.timeout
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
positive integer - only
relevant if the type is set to
‘ring-answer’. Default
value is 2000.
The duration of the ring in milliseconds before the
call is auto answered. If this field is omitted or is left
blank, a value of 2000 is used.
se.rt.x.mod
0,1
Set to 1 if the user interface should allow for
modification by the user of the ringer index used for
this ring class.
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Modification of se.rt.modification.enabledand se.rt.x.nameparameters
through the user interface will be implemented in a future release.
Note
Voice Settings <voice/>
This configuration attribute is defined as follows:
Permitted
Values
Attribute
Default
Interpretation
voice.txPacketFilter
0, 1
Null
Flag to determine whether or not narrowband Tx
high-pass filtering should be enabled.
If set to 1, narrowband Tx high-pass filter is
enabled.
If set 0 or Null, no Tx filtering is performed.
This attribute includes:
•
•
•
•
•
•
•
•
•
•
Voice Coding Algorithms <codecs/>
The following voice codecs are supported:
Sample
Rate
EffectiveAudio
Bandwidth
Algorithm
G.711μ-law
G.711a-law
MIME Type
Label
Bit Rate
64 Kbps
64 Kbps
Frame Size
10ms - 80ms
10ms - 80ms
PMCU
PCMA
G711mu
G711A
8 Ksps
8 Ksps
3.5 KHz
3.5 KHz
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Configuration Files
Sample
Rate
EffectiveAudio
Bandwidth
Algorithm
G.722
MIME Type
G722/8000
G722/16000
Label
G722
G7221
Bit Rate
Frame Size
10ms - 80ms
20ms - 80ms
64 Kbps
16 Ksps
16 Ksps
7 KHz
7 KHz
G.722.1
16 Kbps,
24 Kbps,
32 Kbps
G.729AB
G729
G729AB
8 Kbps
8 Ksps
10ms - 80ms
3.5 KHz
These codecs include:
•
•
Codec Preferences <codecPref/>
Permitted
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 320, 330, 430, 500, 501,
600 and 601 platforms.
1 = highest
3 = lowest
Null = do not use
Give each codec a unique priority, this will
dictate the order used in SDP negotiations.
voice.codecPref.IP_300.G711Mu
voice.codecPref.IP_300.G711A
voice.codecPref.IP_300.G729AB
voice.codecPref.IP_650.G711Mu
voice.codecPref.IP_650.G711A
voice.codecPref.IP_650.G729AB
voice.codecPref.IP_650.G722
voice.codecPref.IP_4000.G711Mu
voice.codecPref.IP_4000.G711A
Null, 1-3
Null, 1-4
1
2
3
2
3
4
1
1
2
Specifies the codec preferences for
SoundPoint IP 301 models. Interpretation
as above.
Specifies the codec preferences for the
SoundPoint IP 550 and 650 platform.
Interpretation as above.
Null, 1-3
Specifies the codec preferences for the
SoundStation IP 4000 platform.
Interpretation as above.
voice.codecPref.IP_4000.G729AB
Null
Not supported by default so that G.711Mu
and G.711A local conferences can be
supported.
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Codec Profiles <audioProfile/>
The following profile attributes can be adjusted for each of the three supported
codecs. In the table, x=G711Mu, G711A, G722, G7221, and G729AB.
Permitted
Attribute
Values
Interpretation
voice.audioProfile.x.payloadSize
10, 20, 30, ...80 Preferred Tx payload size in milliseconds to be
provided in SDP offers and used in the
absence of ptime negotiations. This is also the
range of supported Rx payload sizes.
voice.audioProfile.x.jitterBufferMin
20, 40, 50, 60,
... (multiple of
10)
The smallest jitter buffer depth (in milliseconds)
that must be achieved before play out begins
for the first time. Once this depth has been
achieved initially, the depth may fall below this
point and play out will still continue. This
parameter should be set to the smallest
possible value which is at least two packet
payloads, and larger than the expected short
term average jitter. The IP4000 values are the
same as the IP30x values.
voice.audioProfile.x.jitterBufferShrink
voice.audioProfile.x.jitterBufferMax
10, 20, 30, ...
(multiple of 10)
The absolute minimum duration time (in
milliseconds) of RTP packet Rx with no packet
loss between jitter buffer size shrinks. Use
smaller values (1000 ms) to minimize the delay
on known good networks. Use larger values to
minimize packet loss on networks with large
jitter (3000 ms).
>
The largest jitter buffer depth to be supported
(in milliseconds). Jitter above this size will
always cause lost packets. This parameter
should be set to the smallest possible value
that will support the expected network jitter.
jitterBufferMin,
multiple of 10,
<=300 for IP
320, 330, 430,
501,550, 600,
601, and 650
<= 200 for IP
301
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Configuration Files
Volume Persistence <volume/>
The user’s selection of the receive volume during a call can be remembered
between calls. This can be configured per termination (handset, headset and
hands-free/chassis). In some countries regulations exist which dictate that
receive volume should be reset to nominal at the start of each call on handset
and headset.
Permitted
Attribute
Values
Default
Interpretation
voice.volume.persist.handset
voice.volume.persist.headset
voice.volume.persist.handsfree
0, 1
0
0
1
If set to 1, the receive volume will be
remembered between calls.
0, 1
If set to 0, the receive volume will be reset
to nominal at the start of each call.
0, 1
Gains <gain/>
The default gain settings have been carefully adjusted to comply with the
TIA-810-A digital telephony standard.
Polycom recommends that you do not change these values.
Attribute
Default
voice.gain.rx.analog.handset
voice.gain.rx.analog.headset
voice.gain.rx.analog.chassis
0
0
0
voice.gain.rx.analog.chassis.IP_300
voice.gain.rx.analog.chassis.IP_330
voice.gain.rx.analog.chassis.IP_430
voice.gain.rx.analog.chassis.IP_601
voice.gain.rx.analog.chassis.IP_650
voice.gain.rx.analog.ringer
-6
0
0
6
0
0
voice.gain.rx.analog.ringer.IP_300
voice.gain.rx.analog.ringer.IP_330
voice.gain.rx.analog.ringer.IP_430
voice.gain.rx.analog.ringer.IP_601
-6
0
0
6
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Attribute
Default
0
voice.gain.rx.analog.ringer.IP_650
voice.gain.rx.digital.handset
-15
-21
0
voice.gain.rx.digital.headset
voice.gain.rx.digital.chassis
voice.gain.rx.digital.chassis.IP_330
voice.gain.rx.digital.chassis.IP_430
voice.gain.rx.digital.chassis.IP_4000
voice.gain.rx.digital.chassis.IP_601
voice.gain.rx.digital.chassis.IP_650
voice.gain.rx.digital.ringer
6
6
0
0
6
-21
-12
-12
-21
-21
-12
-14
-24
-24
12
0
voice.gain.rx.digital.ringer.IP_330
voice.gain.rx.digital.ringer.IP_430
voice.gain.rx.digital.ringer.IP_4000
voice.gain.rx.digital.ringer.IP_601
voice.gain.rx.digital.ringer.IP_650
voice.gain.rx.analog.handset.sidetone
voice.gain.rx.analog.handset.sidetone.wideband
voice.gain.rx.analog.headset.sidetone
voice.gain.tx.analog.handset
voice.gain.tx.analog.handset.wideband
voice.gain.tx.analog.headset
3
voice.gain.tx.analog.chassis
3
voice.gain.tx.analog.chassis.IP_300
voice.gain.tx.analog.chassis.IP_330
voice.gain.tx.analog.chassis.IP_430
voice.gain.tx.analog.chassis.IP_601
voice.gain.tx.analog.chassis.IP_650
voice.gain.tx.digital.handset
0
36
36
0
36
0
voice.gain.tx.digital.headset
0
voice.gain.tx.digital.chassis
3
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Configuration Files
Attribute
Default
voice.gain.tx.digital.chassis.IP_330
voice.gain.tx.digital.chassis.IP_430
voice.gain.tx.digital.chassis.IP_4000
voice.gain.tx.digital.chassis.IP_601
voice.gain.tx.digital.chassis.IP_650
voice.gain.tx.analog.preamp.handset
voice.gain.tx.analog.preamp.headset
voice.gain.tx.analog.preamp.chassis
voice.gain.tx.analog.preamp.chassis.IP_601
voice.gain.tx.analog.handset.wideband
voice.handset.rxag.adjust.IP_330
voice.handset.rxag.adjust.IP_430
voice.handset.rxag.adjust.IP_650
voice.handset.txag.adjust.IP_330
voice.handset.txag.adjust.IP_430
voice.handset.txag.adjust.IP_650
voice.handset.sidetone.adjust.IP_330
voice.handset.sidetone.adjust.IP_430
voice.handset.sidetone.adjust.IP_650
voice.handset.wideband
12
12
0
6
12
14
23
32
32
3
1
1
1
9
9
9
0
0
-3
0
voice.handset.wideband.rxdg.adjust
voice.headset.rxag.adjust.IP_330
voice.headset.rxag.adjust.IP_430
voice.headset.rxag.adjust.IP_650
voice.headset.txag.adjust.IP_330
voice.headset.txag.adjust.IP_430
voice.headset.txag.adjust.IP_650
voice.headset.sidetone.adjust.IP_330
voice.headset.sidetone.adjust.IP_430
voice.headset.sidetone.adjust.IP_650
5
4
1
1
21
39
21
-3
-3
-3
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Acoustic Echo Cancellation <aec/>
These settings control the performance of the speakerphone acoustic echo
canceller.
Polycom recommends that you do not change these values.
.
Attribute
Default
voice.aec.hs.enable
0
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
100
7000
-24
-24
-24
voice.aec.hs.erlTab_1500_3500
voice.aec.hs.erlTab_3500_7000
voice.aec.hd.enable
-24
-24
0
voice.aec.hd.lowFreqCutOff
voice.aec.hd.highFreqCutOff
voice.aec.hd.erlTab_0_300
voice.aec.hd.erlTab_300_600
voice.aec.hd.erlTab_600_1500
voice.aec.hd.erlTab_1500_3500
voice.aec.hd.erlTab_3500_7000
voice.aec.hf.enable
100
7000
-24
-24
-24
-24
-24
1
voice.aec.hf.lowFreqCutOff
voice.aec.hf.highFreqCutOff
voice.aec.hf.erlTab_0_300
voice.aec.hf.erlTab_300_600
voice.aec.hf.erlTab_600_1500
voice.aec.hf.erlTab_1500_3500
voice.aec.hf.erlTab_3500_7000
100
7000
-6
-6
-6
-6
-6
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Configuration Files
Acoustic Echo Suppression <aes/>
These settings control the performance of the speakerphone acoustic echo
suppressor.
Polycom recommends that you do not change these values.
Attribute
Default
voice.aes.hs.enable
0
7
0
0
1
7
7
6
6
5
4
4
3
2
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
10
9
8
7
6
5
4
3
2
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Background Noise Suppression <ns/>
These settings control the performance of the transmit background noise
suppression feature.
Polycom recommends that you do not change these values.
Attribute
Default
voice.ns.hs.enable
0
voice.ns.hs.signalAttn
voice.ns.hs.silenceAttn
voice.ns.hd.enable
-6
-9
0
voice.ns.hd.signalAttn
voice.ns.hd.silenceAttn
voice.ns.hf.enable
0
0
1
voice.ns.hf.signalAttn
voice.ns.hf.silenceAttn
voice.ns.hf.IP_4000.enable
voice.ns.hf.IP_4000.signalAttn
voice.ns.hf.IP_4000.silenceAttn
-6
-9
1
-6
-9
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Configuration Files
Automatic Gain Control <agc/>
These settings control the performance of the transmit automatic gain control
feature.
Automatic Gain Control will be implemented in a future release.
Note
Polycom recommends that you do not change these values.
Attribute
Default
voice.agc.hs.enable
voice.agc.hd.enable
voice.agc.hf.enable
0
0
0
Receive Equalization <rxEq/>
These settings control the performance of the receive equalization feature.
Polycom recommends that you do not change these values.
Attribute
Default
voice.rxEq.hs.IP_330.preFilter.enable
voice.rxEq.hs.IP_430.preFilter.enable
voice.rxEq.hs.IP_500.preFilter.enable
voice.rxEq.hs.IP_600.preFilter.enable
voice.rxEq.hs.IP_601.preFilter.enable
voice.rxEq.hs.IP_650.preFilter.enable
voice.rxEq.hs.IP_330.postFilter.enable
voice.rxEq.hs.IP_430.postFilter.enable
voice.rxEq.hs.IP_500.postFilter.enable
voice.rxEq.hs.IP_600.postFilter.enable
voice.rxEq.hs.IP_601.postFilter.enable
1
1
1
1
1
1
0
0
0
0
0
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Attribute
Default
voice.rxEq.hs.IP_650.postFilter.enable
voice.rxEq.hd.IP_330.preFilter.enable
voice.rxEq.hd.IP_430.preFilter.enable
voice.rxEq.hd.IP_500.preFilter.enable
voice.rxEq.hd.IP_600.preFilter.enable
voice.rxEq.hd.IP_601.preFilter.enable
voice.rxEq.hd.IP_650.preFilter.enable
voice.rxEq.hd.IP_330.postFilter.enable
voice.rxEq.hd.IP_430.postFilter.enable
voice.rxEq.hd.IP_500.postFilter.enable
voice.rxEq.hd.IP_600.postFilter.enable
voice.rxEq.hd.IP_601.postFilter.enable
voice.rxEq.hd.IP_650.postFilter.enable
voice.rxEq.hf.IP_330.preFilter.enable
voice.rxEq.hf.IP_430.preFilter.enable
voice.rxEq.hf.IP_500.preFilter.enable
voice.rxEq.hf.IP_600.preFilter.enable
voice.rxEq.hf.IP_601.preFilter.enable
voice.rxEq.hf.IP_650.preFilter.enable
voice.rxEq.hf.IP_4000.preFilter.enable
voice.rxEq.hf.IP_330.postFilter.enable
voice.rxEq.hf.IP_430.postFilter.enable
voice.rxEq.hf.IP_500.postFilter.enable
voice.rxEq.hf.IP_600.postFilter.enable
voice.rxEq.hf.IP_601.postFilter.enable
voice.rxEq.hf.IP_650.postFilter.enable
voice.rxEq.hf.IP_4000.postFilter.enable
0
0
0
0
0
0
1
0
0
0
0
0
0
1
1
1
1
1
1
0
0
0
1
1
1
0
0
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Configuration Files
Transmit Equalization <txEq/>
These settings control the performance of the hands-free transmit equalization
feature.
Polycom recommends that you do not change these values.
.
Attribute
Default
voice.txEq.hs.IP_330.preFilter.enable
voice.txEq.hs.IP_430.preFilter.enable
voice.txEq.hs.IP_500.preFilter.enable
voice.txEq.hs.IP_600.preFilter.enable
voice.txEq.hs.IP_601.preFilter.enable
voice.txEq.hs.IP_650.preFilter.enable
voice.txEq.hs.IP_330.postFilter.enable
voice.txEq.hs.IP_430.postFilter.enable
voice.txEq.hs.IP_500.postFilter.enable
voice.txEq.hs.IP_600.postFilter.enable
voice.txEq.hs.IP_601.postFilter.enable
voice.txEq.hs.IP_650.postFilter.enable
voice.txEq.hd.IP_330.preFilter.enable
voice.txEq.hd.IP_430.preFilter.enable
voice.txEq.hd.IP_500.preFilter.enable
voice.txEq.hd.IP_600.preFilter.enable
voice.txEq.hd.IP_601.preFilter.enable
voice.txEq.hd.IP_650.preFilter.enable
voice.txEq.hd.IP_330.postFilter.enable
voice.txEq.hd.IP_430.postFilter.enable
voice.txEq.hd.IP_500.postFilter.enable
voice.txEq.hd.IP_600.postFilter.enable
voice.txEq.hd.IP_601.postFilter.enable
voice.txEq.hd.IP_650.postFilter.enable
0
0
0
0
0
1
1
1
1
1
1
1
0
0
0
0
0
1
0
0
0
0
0
0
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Attribute
Default
voice.txEq.hf.IP_330.preFilter.enable
voice.txEq.hf.IP_430.preFilter.enable
voice.txEq.hf.IP_500.preFilter.enable
voice.txEq.hf.IP_600.preFilter.enable
voice.txEq.hf.IP_601.preFilter.enable
voice.txEq.hf.IP_650.preFilter.enable
voice.txEq.hf.IP_4000.preFilter.enable
voice.txEq.hf.IP_330.postFilter.enable
voice.txEq.hf.IP_430.postFilter.enable
voice.txEq.hf.IP_500.postFilter.enable
voice.txEq.hf.IP_600.postFilter.enable
voice.txEq.hf.IP_601.postFilter.enable
voice.txEq.hf.IP_650.postFilter.enable
voice.txEq.hf.IP_4000.postFilter.enable
0
0
0
0
0
1
0
1
1
1
1
1
1
0
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Configuration Files
Voice Activity Detection <vad/>
These settings control the performance of the voice activity detection (silence
suppression) feature.
Permitted
Attribute
Values
Default
Interpretation
voice.vadEnable
voice.vadThresh
0, 1
0
If set to 1, enable VAD.
integer from 0
to 30
15
The threshold for determining what is active voice and
what is background noise in dB. This does not apply to
G.729AB codec operation which has its own built-in VAD
function.
voice.vad.signalAnnex 0, 1
B
Null
If set to 1 and voice.vadEnableis set to 1, Annex B is
used. A new line can be added to SDP depending on the
setting of this parameter and the voice.vadEnable
parameter.
•
If voice.vadEnableis set to 1, add attribute line
a=fmtp:18 annexb="yes" below a=rtpmap… attribute
line (where '18' could be replaced by another
payload).
•
If voice.vadEnable is set to 0, add attribute line
a=fmtp:18 annexb="no" below a=rtpmap… attribute
line (where '18' could be replaced by another
payload).
If set to 0 or Null, there is no change to SDP.
Quality of Service <QOS/>
These settings control the Quality of Service (QOS) options.
This attribute includes:
•
•
Ethernet IEEE 802.1p/Q <ethernet/>
The following settings control the 802.1p/Q user_priority field:
•
•
•
Other <other/>
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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.
Call Control <callControl/>
These parameters apply to call control packets, such as the network protocol
signaling.
Permitted
Attribute
Values
Default
Interpretation
qos.ethernet.callControl.user_priority
0-7
5
User-priority used for call control
packets.
Other <other/>
These default parameter values are used for all packets which are not set
explicitly.
Permitted
Attribute
Values
Default
Interpretation
qos.ethernet.other.user_priority
0-7
2
User-priority used for packets that
do not have a per-protocol setting.
IP TOS <IP/>
The following settings control the “type of service” field in outgoing packets:
•
•
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Configuration Files
RTP <rtp/>
These parameters apply to RTP packets.
Permitted
Attribute
Values
Default
Interpretation
qos.ip.rtp.dscp
0 to 63 or
EF or
any of
AF11,AF12,
AF13,AF21,
AF22, AF23,
AF31,AF32,
AF33,AF41,
AF42,AF43
Null
This parameter allows the DSCP of
packets to be specified. If set to a
value, this will override the other
qos.ip.rtp… parameters. Default
of Null which means the other
qos.ip.rtp… parameters will be
used.
qos.ip.rtp.min_delay
0, 1
0, 1
0, 1
0, 1
0-7
1
1
0
0
5
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
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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Call Control <callControl/>
These parameters apply to call control packets, such as the network protocol
signaling.
Permitted
Attribute
Values
Default
Interpretation
qos.ip.callControl.dscp
0 to 63 or
EF or
any of
Null
This parameter allows the DSCP of
packets to be specified. If set to a
value this will override the other
AF11,AF12,
AF13,AF21,
AF22,AF23,
AF31,AF32,
AF33,AF41,
AF42,AF43
qos.ip.callControl…
parameters. Default of Null which
means the other
qos.ip.callControl
…
parameters will be used.
qos.ip.callControl.min_delay
qos.ip.callControl.max_throughput
qos.ip.callControl.max_reliability
qos.ip.callControl.min_cost
0, 1
0, 1
0, 1
0, 1
0-7
1
0
0
0
5
If set to 1, set min-delay bit in the IP
TOS field of the IP header, or else
don’t set it.
If set to 1, set max-throughput bit in
the IP TOS field of the IP header, or
else don’t set it.
If set to 1, set max-reliability bit in
the IP TOS field of the IP header, or
else don’t set it.
If set to 1, set min-cost bit in the IP
TOS field of the IP header, or else
don’t set it.
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.
Basic TCP/IP <TCP_IP/>
This attribute includes:
•
•
•
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Configuration Files
Network Monitoring <netMon/>
Polycom recommends that you do not change these values.
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
tcpIpApp.netMon.enabled
tcpIpApp.netMon.period
0, 1
1
1 to 86400
30
Time Synchronization <sntp/>
The following table describes the parameters used to set up time
synchronization and daylight savings time. The defaults shown will enable
daylight savings time (DST) for North America.
Daylight savings defaults:
•
•
•
Do not use fixed day, use first or last day of week in the month.
Start DST on the second Sunday in March at 2 am.
Stop DST on the first Sunday in November at 2 am.
Permitted
Attribute
Values
Default
Interpretation
tcpIpApp.sntp.resyncPeriod
positive
integer
86400 (24
hours)
Time in seconds between
Simple Network Time
Protocol (SNTP) re-syncs.
tcpIpApp.sntp.address
valid host
name or IP
address
clock
Address of the SNTP
server.
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Permitted
Values
Attribute
Default
Interpretation
tcpIpApp.sntp.address.overrideDHCP
0, 1
0
These parameters
determine whether
configuration file
parameters override DHCP
parameters for the SNTP
server address and
Greenwich Mean Time
(GMT) offset. If set to 0,
DHCP values will override
configuration file
parameters. If set to 1, the
configuration file
parameters will override
DHCP values.
tcpIpApp.sntp.gmtOffset
positive or
negative
integer
-28800
(Pacific
time)
Offset in seconds of the
local time zone from GMT.
3600 seconds = 1 hour
tcpIpApp.sntp.gmtOffset.overrideDHCP
0, 1
0
These parameters
determine whether
configuration file
parameters override DHCP
parameters for the SNTP
server address and GMT
offset. If set to 0, DHCP
values will override
configuration file
parameters. If set to 1, the
configuration file
parameters will override
DHCP values.
tcpIpApp.sntp.daylightSavings.enable
0, 1
0, 1
1
0
If set to 1, apply daylight
savings rules to displayed
time.
tcpIpApp.sntp.daylightSavings.fixedDayEnable
If set to 0, month, date, and
dayOfWeek are used in
DST date calculation.
If set to 1, then only month
and date are used.
tcpIpApp.sntp.daylightSavings.start.month
1-12
3 (March)
Month to start DST.
Mapping: 1=Jan, 2=Feb, ...,
12=Dec
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Configuration Files
Permitted
Values
Attribute
Default
Interpretation
tcpIpApp.sntp.daylightSavings.start.date
1-31
8
If fixedDayEnableis set to
1, use as day of the month
to start DST.
If fixedDayEnableis set to
0, us the mapping: 1 = the
first occurrence of a given
day-of-the-week in a month,
8 = the second occurrence
of a given day-of-the-week
in a month, 15 = the third
occurrence of a given
day-of-the-week in a month,
22 = the fourth occurrence
of a given day-of-the-week
in a month
tcpIpApp.sntp.daylightSavings.start.time
0-23
1-7
2
1
0
Time of day to start DST in
24 hour clock.
Mapping: 2=2 am, 14=2 pm
tcpIpApp.sntp.daylightSavings.start.dayOfWeek
Day of week to apply DST.
Mapping: 1=Sun, 2=Mon,
..., 7=Sat
tcpIpApp.sntp.daylightSavings.start.dayOfWeek.la
stInMonth
0, 1
If set to 1 and
fixedDayEnable is set to 0,
DST starts on the last day
(specified by
start.dayOfWeek) of the
week in the month. The
start.dateis ignored.
tcpIpApp.sntp.daylightSavings.stop.month
tcpIpApp.sntp.daylightSavings.stop.date
1-12
1-31
11
1
Month to stop DST.
Day of the month to stop
DST.
tcpIpApp.sntp.daylightSavings.stop.time
0-23
2
Time of day to stop DST in
24 hour clock.
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek
1-7
1
0
Day of week to stop DST.
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.la
stInMonth
0, 1
If set to 1 and
fixedDayEnable set to 0,
DST stops on the last day
(specified by
stop.dayOfWeek) of the
week in the month. The
stop.dateis ignored.
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Port <port/>
This attribute includes:
•
RTP <rtp/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
tcpIpApp.port.rtp.filterByIp
0, 1
1
If set to 1, reject RTP packets
arriving from (sent from) a
non-negotiated (through SDP) IP
address.
tcpIpApp.port.rtp.filterByPort
tcpIpApp.port.rtp.forceSend
0, 1
0
If set to 1, reject RTP packets
arriving from (sent from) a
non-negotiated (through SDP)
port.
Null,
1024-65534
Null
When non-Null, send all RTP
packets to, and expect all RTP
packets to arrive on, the
specified port.
Note: both
tcpIpApp.port.rtp.filterByIp and
tcpIpApp.port.rtp.filterByPort
must be enabled for this to work.
tcpIpApp.port.rtp.mediaPortRangeStart
Null, even
integer from
1024-65534
Null
If set to Null, the value 2222 will
be used for the first allocated
RTP port, otherwise, the
specified port will be used.
Subsequent ports will be
allocated from a pool starting
with the specified port plus two
up to a value of (start-port + 46),
after which the port number will
wrap back to the starting value.
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.
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Configuration Files
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
httpd.enabled
0, 1
1
If set to 1, the HTTP server will be enabled.
This attribute also includes:
Configuration <cfg/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
httpd.cfg.enabled
0, 1
1
If set to 1, the HTTP server configuration interface
will be enabled.
httpd.cfg.port
1-65535
80
Port is 80 for HTTP servers. Care should be taken
when choosing an alternate port.
Call Handling Configuration <call/>
This configuration attribute is defined as follows:
Permitted
Values
Attribute
Default
Interpretation
call.rejectBusyOnDnd
0, 1
1
If set to 1, reject all incoming calls with the
reason “busy” if do-not-disturb is enabled.
Note: This attribute is ignored when the line is
configured as shared. The reason being that
even though one party has turned on DND, the
other person/people sharing that line do not
necessarily want all calls to that number diverted
away.
Note: If server-based DND is enabled, this
parameter is disabled.
call.enableOnNotRegistered
0, 1
1
If set to 1, calls will be allowed when the phone is
not successfully registered, otherwise, calls will
not be permitted without a valid registration.
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Permitted
Attribute
Values
Default
Interpretation
call.offeringTimeOut
positive
integer
60
Time in seconds to allow an incoming call to ring
before dropping the call, 0=infinite.
Note: The call diversion, no answer feature will
take precedence over this feature if enabled. For
more information, refer to No Answer
call.ringBackTimeOut
call.dialtoneTimeOut
positive
integer
60
60
Time in seconds to allow an outgoing call to
remain in the ringback state before dropping the
call, 0=infinite.
Null, positive
integer
Time in seconds to allow the dialtone to be
played before dropping the call.
If set to 0, the call is not dropped.
If set to Null, call dropped after 60 seconds.
call.lastCallReturnString
call.callsPerLineKey
string of
maximum
length 32
*69
The string sent to the server when the user
selects the “last call return” action.
1 to 24 OR
1 to 8
24 OR
8
For the SoundPoint IP 600, 601, and 650, the
permitted range is 1 to 24 and the default is 24.
For all other phones the permitted range is 1 to 8
and the default is 8.
This is the number of calls that may be active or
on hold per line key on the phone.
Note that this may be overridden by the
per-registration attribute of
call.stickyAutoLineSeize
Null, 0, 1
0
If set to 1, makes the phone use "sticky" line
seize behavior. This will help with features that
need a second call object to work with. The
phone will attempt to initiate a new outgoing call
on the same SIP line that is currently in focus on
the LCD (this was the behavior in SIP 1.6.5).
If set to 0 or Null, the feature is disabled (this was
the behavior in SIP 1.6.6).
Note: This may fail due to glare issues in which
case the phone may select a different available
line for the call.
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Configuration Files
Permitted
Values
Attribute
Default
Interpretation
call.stickyAutoLineSeize.onHook
Dialing
Null, 0, 1
Null
If call.stickyAutoLineSeizeis set to 1, this
parameter has no effect. The regular
stickyAutoLineSeize behavior is followed.
If call.stickyAutoLineSeizeis set to 0 or Null
and this parameter is set to 1, this overrides the
stickyAutoLineSeize behavior for hot dial only.
(Any new call scenario seizes the next available
line.)
If call.stickyAutoLineSeizeis set to 0 or Null
and this parameter is set to 0 or Null, there is no
difference between hot dial and new call
scenarios.
Note: A hot dial occurs on the line which is
currently in the call appearance. Any new call
scenario seizes the next available line.
call.transfer.blindPreferred
0,1
Null
If set to 1, the Blind soft key appears as a
transfer type.
If set to 0 or Null, the Normal soft key appears.
Note: This parameter is supported on the
SoundPoint IP 330/320 only.
This attribute also includes:
•
•
Shared Calls <shared/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
call.shared.disableDivert
0, 1
1
If set to 1, disable diversion feature for shared
lines.
Note: This feature is disabled on most call
servers.
call.shared.seizeFailReorder
0, 1
1
If set to 1, play re-order tone locally on shared
line seize failure.
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Permitted
Values
Attribute
Default
Interpretation
call.shared.oneTouchResume
0, 1
0
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.
Note: This parameter affects the SoundStation
IP 4000 phone. For other phones a quick press
and release of the line key will resume a call
whereas pressing and holding down the line
key will show a list of calls on that line.
call.shared.exposeAutoHolds
0, 1
0
If set to 1, on a shared line, when setting up a
conference, a re-INVITE will be sent to the
server.
If set to 0, no re-INVITE will be sent to the
server.
Hold, Local Reminder <hold/><localReminder/>
This configuration attribute is defined as follows:
Permitted
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
integer
60
90
Time in seconds between subsequent
reminders.
call.hold.localReminder.startDelay
non-negative
integer
Time in seconds to wait before the
initial reminder.
Directory <dir/>
The directory is stored in either flash memory or RAM on the phone. The
directory size is limited based on the amount of flash memory in the phone.
(Different phone models have variable flash memory.)
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Configuration Files
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
Attribute
Values
Default
Interpretation
dir.local.volatile.2meg
0, 1
0
Attribute applies to platforms with 2
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.2meg
dir.local.volatile.4meg
1 to 20
20
0
Attribute applies to platforms with 2
Mbytes of flash memory. Maximum
size in Kbytes of non-volatile
storage that the directory will be
permitted to consume.
0, 1
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 directory will be
permitted to consume.
Note: For the SoundPoint IP 650
platform, this value is internally
replaced by 2X the value.
dir.local.volatile.8meg
0, 1
0
Attribute applies only to platforms
with 8 Mbytes of flash memory.
If set to 1, use volatile storage for
phone-resident copy of the directory
to allow for larger size.
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Permitted
Values
Attribute
Default
Interpretation
dir.local.nonVolatile.maxSize.8meg
1 to 100
100
Attribute applies only to platforms
with 8 Mbytes of flash memory.
This is the maximum size of
non-volatile storage that the
directory will be permitted to
consume.
dir.search.field
0, 1
Null
Specifies how to search the contact
directory. If set to 1, search by
contact’s first name. If set to 0,
search by contact’s last name.
Presence <pres/>
The parameter pres.regis the line number used to send SUBSCRIBE. If this
parameter is missing, the phone will use the primary line to send SUBSCRIBE.
Permitted
Attribute
Values
Default
Interpretation
pres.reg
positive
integer
1
Specifies the line/registration
number used to send SUBSCRIBE
for presence. Must be a valid
line/registration number. If the
number is not a valid
line/registration number, it is
ignored.
Fonts <font/>
This section does not apply to the SoundPoint IP 301 phones.
Note
These settings control the phone’s ability to dynamically load an external font
file during 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 ranges not already embedded. The font file must be
a Microsoft .fnt or .fon file format. (.fon file format is a collection of .fnt fonts
grouped together within a single file.) The font file name must follow a specific
pattern as described:
•
Font filename:
<fontName>_<fontHeightInPixels>_<fontRange>.<fontExtension>
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•
<fontName> is a free string of characters that typically carries the meaning
of the font. Examples are “fontFixedSize” for a fixed-size font, or
“fontProportionalSize” for a proportional size font.
•
•
<fontHeightInPixels> describes the font height in number of screen pixels.
<fontRange> describes the Unicode range covered by this font. Since .fnt
or .fon are 256 characters based blocks, the <fontRange> is Uxx00_UxxFF
(.fnt file) or Uxx00_UyyFF (.fon file). For more information, refer to
•
<fontExtension> describes the file type. Either .fnt for single 256
characters font or .fon for multiple .fnt files.
If it is necessary to overwrite an existing font, use these
<fontName>_<fontHeightInPixels>:
SoundPoint IP 320, 330, 430, 500 and 501
“fontProp_10”
This is the font used widely in the current implementation.
This is the soft key specific font.
“fontPropSoftkey_10”
SoundPoint IP 550, 600, 601, and 650
“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 SoundPoint 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. For example:
•
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
currently supported. For example, for the Eastern/Central European
Czech language, 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 different scrambled .fnt blocks. This font delimiter must be placed
in the “copyright” attribute of the .fnt header. .fon files are useful if you want
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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.
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
font.delimiter
string up to 256 ASCII
characters
Null
Delimiter required to retrieve different
grouped .fnt blocks.
This attribute also includes:
•
•
•
•
IP_330 font <IP_330/>
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
font.IP_330.x.name
fontName_height_Uxx00_U Null
yyFF.fon OR
Defines the font file that will be loaded from
boot server during boot up.
fontName_height_Uxx00_U
xxFF.fnt
Note: When several font.IP_330.x.name
are defined, the index x must follow
consecutive increasing order.
IP_400 font <IP_400/>
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
font.IP_400.x.name
fontName_height_Uxx00_U Null
yyFF.fon OR
Defines the font file that will be loaded from
boot server during boot up.
fontName_height_Uxx00_U
xxFF.fnt
Note: When several font.IP_430.x.name
are defined, the index x must follow
consecutive increasing order.
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IP_500 font <IP_500/>
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
font.IP_500.x.name
fontName_height_Uxx00_U Null
yyFF.fon OR
Defines the font file that will be loaded from
boot server during boot up.
fontName_height_Uxx00_U
xxFF.fnt
Note: When several font.IP_500.x.name
are defined, the index x must follow
consecutive increasing order.
IP_600 font <IP_600/>
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
font.IP_600.x.name
fontName_height_Uxx00
_UyyFF.fon OR
Null
Defines the font file that will be loaded from
boot server during boot up.
fontName_height_Uxx00
_UxxFF.fnt
Note: When several font.IP_600.x.name
are defined, the index x must follow
consecutive increasing order.
Keys <key/>
These settings control the scrolling behavior of keys and can be used to change
key functions.
Permitted
Attribute
Values
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-specific soft keys. The value is
an integer multiple of 500 milliseconds (1=500ms).
SoundPoint IP 301, 320, 330, 430, 501, 550, 600, 601, and 650 and SoundStation
IP 4000 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 the
<keys .../> element of the configuration file. These will override the built-in
assignments.
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Polycom does not recommend the remapping for keys.
In the following table, x=IP_300, IP_330, IP 430, IP_500, IP_550, IP_600, IP_650,
IP_4000, and IP_7000 and y is the key number. Note that IP_300 parameters
affect SoundPoint IP 301 phones, IP_330 parameters affect SoundPoint IP 320
and 330 phones, IP_430 parameters affect SoundPoint IP 430 phones, IP_500
parameters affect SoundPoint IP 501 phones, IP_550 parameters affect
SoundPoint IP 550 phone, IP_600 parameters affect SoundPoint IP 600 and 601
phones, IP_650 parameters affect SoundPoint IP 650 phones, IP_4000
parameters affect the SoundStation IP 4000 phones. IP 300: y=1-35; IP 330:
y=1-34; IP 430: y=1-35; IP 500: y=1-40; IP_550: y=1-40; IP 600: y=1-42;
IP_650:y=1-42; IP_4000:y=1-29.
Attribute
Permitted Values
Interpretation
key.x.y.function.prim
key.x.y.subPoint.prim
Functions listed below. Sets the function for key y on platform x.
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:
Functions
ArrowDown
ArrowLeft
ArrowRight
ArrowUp
BuddyStatus
CallList
Dialpad5
Dialpad6
Dialpad7
Dialpad8
Dialpad9
DialpadStar
DialpadPound
Directories
DoNotDisturb
Handsfree
Headset
Line2
Select
Line3
Setup
Line4
SoftKey1
SoftKey2
SoftKey3
SoftKey4
SpeedDial
SpeedDialMenu
Transfer
VolDown
VolUp
Line5
Line6
Messages
Menu
Conference
Delete
MicMute
MyStatus
Null
Dialpad0
Dialpad1
Dialpad2
Dialpad3
Dialpad4
Offline
Redial
Release
Hold
Line1
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Configuration Files
Bitmaps <bitmap/>
The bitmaps used by each phone model are defined in this section.
Platform <IP_300/>, <IP 330/>, <IP_400/>, <IP_500/>, <IP_600/>,
<IP_4000/>
In the following table, x=IP_300, IP_330, IP_400, IP_500, IP_600, IP_4000 and y
is the bitmap number. Note that IP_300 parameters affect SoundPoint IP 301
phones, IP_330 parameters affect SoundPoint IP 320 and 330 phones, IP_400
parameters affect SoundPoint IP 430 phones, IP_500 parameters affect
SoundPoint IP 501 phones, IP_600 parameters affect SoundPoint IP 550, 600,
601, and 650 phones, IP_4000 parameters affect SoundStation IP 4000 phones.
Attribute
Permitted Values
Interpretation
bitmap.x.y.name
The name of a bitmap
to be used.
This is the name of a bitmap to be used for creating an
animation. If the bitmap is to be downloaded from the boot
server, its name must:
•
•
Be different from any name already in use in sip.cfg.
Match the name of the corresponding <fileName>.bmp to
be retrieved from the boot server.
Indicators <ind/>
The following indicators are used by the phone:
•
•
•
•
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This configuration attribute is defined as follows:
Permitted
Values
Attribute
Default
Interpretation
ind.idleDisplay.mode
1 (default), 2,
3
Null
The idle display animation screen layouts.
For example, for the SoundPoint IP 330/320:
•
•
•
If set to 1 or Null, the idle display animation
size is 87 x 11 pixels.
If set to 2, the idle display animation size is 87
x 22 pixels.
If set to 3, the idle display animation size is
102 x 22 pixels.
ind.idleDisplay.enabled
0, 1
0
If set to 1, the idle display may support
presentation of a custom animation if configured
in the animation section of sip.cfg.
Animations <anim/> <IP_300/>, <IP_330/>, <IP_400/>, <IP_500/>,
<IP_600/>, <IP_4000/>
This section defines bitmap animations composed of bitmap/duration
couples. In the following table, x=IP_300, IP_330, IP_400, IP_500, IP_600,
IP_4000 , y is the animation number, z is the step in the animation. Note that
IP_300 parameters affect SoundPoint IP 301 phones, IP_330 parameters affect
SoundPoint IP 320 and 330 phones, IP_400 parameters affect SoundPoint IP
430 phones, IP_500 parameters affect SoundPoint IP 501 phones and IP_600
parameters affect SoundPoint IP 550, 600, 601, and 650 phones, IP_4000
parameters affect SoundStation IP 4000 phones.
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, refer to
<IP_500/>, <IP_600/>, <IP_4000/> on page
A-65.
ind.anim.x.y.frame.z.duration
positive integer
Duration in milliseconds for this step. 0=infinite.
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Configuration Files
Patterns <pattern/>
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
Attribute
Values
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.
Duration in milliseconds for this step. 0=infinite
For bi-color LEDs, specify color.
positive integer
Red or Green
(default is Red if
not specified)
Classes <class/>
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 number for that class.
Permitted
Attribute
Values
Interpretation
ind.class.x.state.y.index
positive integer
For LED type indicators, index refers to the pattern index,
For Graphic Icon type indicators, index refers to the
animation index, such as index y in the Animations <anim/>
<IP_4000/> tag above.
Assignments
This attribute assigns a type and a class to an indicator. In the case of the
Graphic Icon type, it also assigns a physical location and size in pixels on the
LCD display (refer to the following section). In the case of the LED type, it
assigns a physical LED number (refer to Graphic Icons <gi/> <IP_300/>,
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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
this indicator.
ind.led.x.physNum
This maps the logical index to a specific physical LED.
Graphic Icons <gi/> <IP_300/>, <IP_330>, <IP_400/>, <IP_500/>, <IP_600/>,
<IP_4000/>
In the following table, x=IP_300, IP_330, IP_400, IP_500, IP_600, IP_4000, y is
the graphic icon number. Note that IP_300 parameters affect SoundPoint IP
301 phones, IP_330 parameters affect SoundPoint IP 320 and 330 phones,
IP_400 parameters affectSoundPoint IP 430 phones, IP_500 parameters affect
SoundPoint IP 501 phones, and IP_600 parameters affect SoundPoint IP 550,
600, 601, and 650 phones, IP_4000 parameters affect SoundStation IP 4000
phones.
Permitted
Attribute
Values
Interpretation
ind.gi.x.y.index
This is for internal usage only and should not be changed (this is
the logical index).
ind.gi.x.y.class
ind.gi.x.y.physX
positive integer
this indicator.
IP 300: 0-19
For Graphic Icon type indicators, this is the x-axis location of the
upper left corner of the indictor measured in pixels from left to
right.
IP 330: 0-101
IP 400: 0-122
IP 500: 0-159
IP 600: 0-319
IP 4000: 0-247
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Configuration Files
Permitted
Values
Attribute
Interpretation
ind.gi.x.y.physY
IP 300: 0-3
For Graphic Icon type indicators, this is the y-axis location of the
upper left corner of the indicator measured in pixels from top to
bottom.
IP 330: 0-19
IP 400: 0-45
IP 500: 0-79
IP 600: 0-159
IP 4000: 0-67
ind.gi.x.y.physW
IP 300: n/a
For Graphic Icon type indicators, this is the width of the indicator
measured in pixels.
IP 330: 1-87
IP 400: 1-102
IP 500: 1-160
IP 600: 1-320
IP 4000: 1-248
ind.gi.x.y.physH
IP 300: n/a
For Graphic Icon type indicators, this is the height of the indicator
measured in pixels.
IP 330: 1-20
IP 400: 1-23
IP 500: 1-80
IP 600: 1-160
IP 4000: 1-68
Event Logging <log/>
Logging parameter changes can impair system operation. Do not change any
logging parameters without prior consultation with Polycom Technical Support.
Caution
The event logging system supports the following classes of events:
Level
Interpretation
0
1
2
3
4
5
6
Debug only
High detail event class
Moderate detail event class
Low detail event class
Minor error - graceful recovery
Major error - will eventually incapacitate the system
Fatal error
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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
Two types of logging are supported:
•
•
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Configuration Files
Basic Logging <level/><change/> and <render/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
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.
Possible values for xxx are so,
app1, sip, sspsc, ssps, pps, net,
cfg, cdp, pmt, ftp, ares, dns, cxss,
httpd, rdisk, copy, slog, res, key,
log, curl, rtos, mb, ib, sotet, ttrs,
srtp, usb, .
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 memory-based log system.
The log.render.levelmaps to
syslog severity as follows:
0 -> SeverityDebug (7)
1 -> SeverityDebug (7)
2 -> SeverityInformational (6)
3 -> SeverityInformational (6)
4 -> SeverityError (3)
5 -> SeverityCritical (2)
6 -> SeverityEmergency (0)
7 -> SeverityNotice (5)
For more information, refer to
log.render.type
0-2
2
1
Refer to above table for timestamp
type.
log.render.realtime
0, 1
Set to 1.
Note: Polycom recommends that
you do not change this value.
log.render.stdout
log.render.file
0, 1
0, 1
1
1
Set to 1.
Note: Polycom recommends that
you do not change this value.
Set to 1.
Note: Polycom recommends that
you do not change this value.
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Permitted
Values
Attribute
Default
Interpretation
log.render.file.size
positive
integer, 1 to
179.5
16
Maximum local application log file
size in Kbytes. When this size is
exceeded, the file is uploaded to
the boot server and the local copy
is erased.
log.render.file.upload.period
log.render.file.upload.append
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.
0, 1
1
If set to 1, use append mode when
uploading log files to server.
Note: HTTP and TFTP don’t
support append mode unless the
server is set up for this.
log.render.file.upload.append.sizeLimit
log.render.file.upload.append.limitMode
positive
integer
512
Maximum log file size on boot
server in Kbytes.
delete, stop
delete
Behavior when server log file has
reached its limit.
delete=delete file and start over
stop=stop appending to file
Scheduled Logging Parameters <sched/>
The phone can be configured to schedule certain advanced logging tasks on a
periodic basis. These attributes should be set in consultation with Polycom
Technical Support. Each scheduled log task is controlled by a unique attribute
set starting with log.sched.x where x identifies the task.
Permitted
Attribute
Values
Interpretation
log.sched.x.name
alphanumeric
string
Name of an internal system command to be periodically 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
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Configuration Files
Permitted
Values
Attribute
Interpretation
log.sched.x.startMode
log.sched.x.startTime
abs, rel
Start at absolute time or relative to boot.
positive
Seconds since boot when startMode is rel or the start time in 24-hour
integer OR
hh:mm
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
Security <sec/>
This attribute’s settings affect security aspects of the phone.
This configuration attribute is defined as follows:
.
Permitted
Attribute
Values
Default
Interpretation
sec.tagSerialNo
0, 1
Null
If set to 1, the phone may advertise its serial number
(Ethernet address) through protocol signaling.
If set to 0 or Null, the phones does advertise its serial
number.
This attribute also includes:
•
•
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Encryption <encryption/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
sec.encryption.upload.dir
0, 1
0
If set to 0, the phone-specific contact directory is
uploaded to the server unencrypted regardless of
how it was downloaded. This will replace whatever
phone-specific contact directory is on the server
even if it is encrypted.
If set to 1, the phone-specific contact directory is
uploaded encrypted regardless of how it was
downloaded. This will replace whatever
phone-specific contact directory is on the server
even if it is unencrypted.
sec.encryption.upload.ove
rrides
0, 1
0
If set to 0, the phone-specific configuration override
file (<Ethernet Address>-phone.cfg) is uploaded
unencrypted regardless of how it was downloaded.
This will replace the override file on the server
even if it is encrypted.
If set to 1, the phone-specific configuration override
file is uploaded encrypted regardless of how it was
downloaded. This will replace the override file on
the server even if it is unencrypted.
Password Lengths <pwd/><length/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
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
License <license/>
This attribute’s settings control aspects of the feature licensing system.
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
license.polling.time
00:00 – 23:59
2:00am
The time to check whether or not the license has
expired.
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Configuration Files
Provisioning <prov/>
This attribute’s 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
prov.fileSystem.ffs0.8meg.minFreeSpace
5-512
Minimum free space in Kbytes to
reserve in the file system when
downloading files from the boot
server.
420
48
Note: Polycom recommends that
you do not change these
parameters.
512
Note: For the SoundPoint IP 650
platform,
prov.fileSystem.ffs0.8meg.m
inFreeSpace is internally
replaced by 2X the value.
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. Polling
time.
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RAM Disk <ramdisk/>
This attribute’s settings control the phone’s internal RAM disk feature.
Polycom recommends that you do not change these values.
Permitted
Attribute
Values
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.
ramdisk.bytesPerBlock
ramdisk.blocksPerTrack
ramdisk.nBlocks
0, 32, 33, ...,
1024
0
These three parameters use internal defaults
when value is set to 0.
Note: For the SoundPoint IP 650 platform,
ramdisk.bytesPerBlock is internally replaced by 2X
the value.
0, 1, 2, ...,
65536
0
0, 1, 2, ...,
65536
4096
50
ramdisk.minsize
50 to 16384
Smallest size in Kbytes of RAM disk to create
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.
Request <request/>
This attribute includes:
•
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Configuration Files
Delay <delay/>
These settings control the phone’s behavior when a request for restart or
reconfiguration is received.
Permitted
Attribute
Values
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 executed 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.
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
“presence”,
Interpretation
feature.x.name
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 feature (the
“calllist” feature must be enabled for this feature to be
available)
“call-park”,
“group-call-pickup”,
“directed-call-pickup”,
“last-call-return”,
“acd-login-logout”,
“acd-agent-available”
“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)
“url-dialing” controls whether URL/name dialing is available
from a private line (it is never available from a shared line)
“call-park” is the call park and park-retrieve features
“group-call-pickup” is the group call pickup feature
“directed-call-pickup” is the directed call pickup feature
“last-call-return” is the last call return feature
“acd-login-logout” is the ACD login/logout feature
“acd-agent-available” is the ACD agent
available/unavailable feature
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Attribute
Permitted Values
Interpretation
feature.x.enabled
0 or 1 (default) except for
x=9
If set to 0, the feature will be disabled.
If set to 1, the feature will be enabled and usable by the local
user.
Note: The "url-dialing" feature must be disabled by setting
feature.9.enabled to 0 in order to prevent unknown
callers from being identified on the display by an IP address.
Note: The “call list” feature can be disabled on all
SoundPoint IP and SoundStation IP platforms except the
SoundPoint IP 330/320.
Resource <res/>
This attribute’s settings control the maximum size or an external resource
retrieved at run time.
This attribute also includes:
•
•
Finder <finder/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
res.finder.sizeLimit
positive
integer
300
If a resource that is being downloaded to the phone
is larger than this value * 1024 bytes (= the
maximum size), the resource will be automatically
truncated to the maximum size defined.
Note: For the SoundPoint IP 650 platform, this value
is internally replaced by 2X the value.
res.finder.minfree
1 to 2048
600
A resource will not be downloaded to the phone if the
amount of free memory is less than this value * 1024
bytes (= the minimum size). This parameter is used
for 16MB SDRAM platforms and scaled up for
platforms with more SDRAM.
If set to 0 or Null, the default value of 600 is used.
Note: For the SoundPoint IP 650 platform, this value
is internally replaced by 2X the value.
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Configuration Files
Quotas <quotas/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Interpretation
res.quotas.x.name
“tone”=1, “bitmap”,
or “font”
The name of the sub-application for which the particular quota
will apply:
“tone” relates to all downloaded tones and sound effects
“bitmap” relates to all downloaded bitmaps
“font” relates to all downloaded fonts
res.quotas.x.value
positive integer
When a particular resource (one of category “font”, “bitmap”, or
“font”) is 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 exceed the quota
for that category, the resource will not be downloaded and a
predefined default will be used instead.
For res.quotas.x.value, the default is 600 KB for tones and
10 KB for bitmaps and fonts.
Note: For the SoundPoint IP 650 platform, this value is
internally replaced by 2X the value.
Microbrowser <mb/>
This attribute’s settings control the home page, proxy and size limits to be used
by the Microbrowser when it is selected to provide services. The Microbrowser
is supported on the SoundPoint IP 430, 501, 550, 601, and 650 and the
SoundStation IP 4000 phones.
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
mb.proxy
Null or
Null.
Address of the desired HTTP proxy to be used
by the Microbrowser. If blank, normal unproxied
HTTP is used by the Microbrowser.
domain name or
IP address in the
format
Default
port =
8080
<address>:<port>
This attribute also includes:
•
•
•
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Idle Display <idleDisplay/>
The Microbrowser can be used to create a display that will be part of the
phone’s idle display. These settings control the home page and the refresh rate.
Attribute
Permitted Values
Default
Interpretation
mb.idleDisplay.home
Null or any fully
Null
URL used for Microbrowser idle display home
page. For example:
formed valid HTTP
URL. Length up to
255 characters.
http://www.example.com/xhtml/frontpage.cgi?pa
ge=home. If empty, there will be no
Microbrowser idle display feature. Note that the
Microbrowser idle display will displace the idle
display indicator (refer to
mb.idleDisplay.refresh
0 or an integer > 5
0
The period in seconds between refreshes of the
idle display Microbrowser's content. If set to 0,
the idle display Microbrowser is not refreshed.
The minimum refresh period is 5 seconds
(values from 1 to 4 are ignored, and 5 is used).
Note: If an HTTP Refresh header is detected, it
will be respected, even if this parameter is set to
0. The refresh parameter will be respected only
in the event that a refresh fails. Once a refresh is
successful, the value in the HTTP refresh
header, if available, will be used.
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Configuration Files
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.
For example:
http://www.example.com/xhtml/frontpage.cgi?pa
ge=home.
mb.main.statusbar
0, 1
Null
Null
Flag to determine whether or not to turn off
display of status messages.
If set to 1, the display of the status bar is
enabled.
If set to 0, or Null, the display of the status bar is
disabled.
mb.main.idleTimeout
0 - 10, minutes
Timeout for the interactive browser. If the
interactive browser remains idle for a defined
period of time, the phone should return to the
idle browser.
If set to 0 or Null, there is no timeout.
If set to value greater than 0 and less than 10,
the timeout is for that number of minutes.
Browser Limits <limits/>
These settings limit the size of object which the Microbrowser will display by
limiting the amount of memory available for the Microbrowser.
Attribute
Permitted Values
Default
Interpretation
mb.limits.nodes
positive integer
256
Limits the number of tags that the XML parser
will handle. This limits the amount of memory
used by complicated pages. A maximum total of
500 (256 each) is recommended. This value is
used as referent values for 16MB of SDRAM.
Note: Increasing this value may have a
detrimental effect on performance of the phone.
mb.limits.cache
positive integer
200
Limits the total size of objects downloaded for
each page (both XHTML and images). Once this
limit is reached, no more images are
downloaded until the next page is requested.
Units = kBytes. This value is used as referent
values for 16MB of SDRAM.
Note: Increasing this value may have a
detrimental effect on performance of the phone.
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USB Port <usb/>
This attribute’s settings control the USB port.
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
usb.enable
0, 1
Null
This parameter enables or disables the USB
port on the phone.
This attribute also includes:
•
Bulk Drive <bulkDrive/>
These settings control the bulk drive or memory stick.
Attribute
Permitted Values
Default
Interpretation
usb.bulkDrive.enable
0, 1
Null
This parameter enables or disables support for a
USB bulk drive connected to the USB port on
the phone.
usb.bulkDrive.name
alphanumeric string
usbDrive
This parameter is a string which specifies the
name of the mounted USB drive.
Per-Phone Configuration
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
configuration files. For more information, refer to Deploying Phones From the
Polycom recommends that you create another file with your organization’s
modifications. If you must change any Polycom templates, back them up first.
The parameters include:
•
•
•
•
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Configuration Files
•
•
•
•
•
Registration <reg/>
SoundPoint IP 301, 320, 330, and 430 support a maximum of two unique
registrations, SoundPoint IP 501 supports three, the SoundPoint IP 550
supports four, and SoundPoint IP 600, 601, and 650 support six. Up to three
SoundPoint IP Expansion Modules can be added to a single host SoundPoint
IP 601 and 650 phone increasing the total number of buttons to 12 registrations
on the IP 601 and 34 registrations on the IP 650. Each registration can
optionally be associated with a private array of servers for completely
segregated signaling. The SoundStation IP 4000 supports a single registration.
In the following table, x is the registration number. IP 301, 320, 330, 430: x=1-2;
IP 501: x=1-3; IP 550: x=1-4; IP 600: x=1-6; IP 601: x=1-12;
IP 650: x=1-34; IP 4000: x=1.
Permitted
Attribute
Values
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 format
userPart from
Null
The user part or the user and the host part of
the phone’s SIP URI.
userPart@domain
The user part of the phone's SIP URI. For
example, reg.x.address=”1002” from
reg.x.address=”[email protected]”.
reg.x.label
UTF-8 encoded
string
Null
Text label to appear on the display adjacent
to the associated line key. If omitted, the label
will be derived from the user part of
reg.x.address.
reg.x.lcs
0, 1
0
If set to 1, the Microsoft Live Communications
Server is supported for registration x.
reg.x.type
private OR shared
private
If set to private, use standard call signaling.
If set to shared, augment call signaling with
call state subscriptions and notifications and
use access control for outgoing calls.
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Permitted
Attribute
Values
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 appearance (BLA). It must
be Null in all other cases.
reg.x.auth.userId
string
Null
Null
User ID to be used for authentication
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
string
Password to be used for authentication
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
Null
Null
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 may override the servers
reg.x.server.y.port
0, Null, 1 to 65535
reg.x.server.y.transport
DNSnaptr or
TCPpreferred or
UDPOnly or
TLS or
DNSnap
tr
Note: If the reg.x.server.y.address parameter
is non-Null, all of the reg.x.server.y.xxx
parameters will override the parameters
TCPOnly
reg.x.server.y.expires
positive integer
0, 1
Null
Null
60
reg.x.server.y.register
Note: If the reg.x.server.y.address parameter
is non-Null, it takes precedence even if the
DHCP server is available.
reg.x.server.y.expires.overlap
positive integer,
minimum 5,
maximum 65535
Note: TLS is not supported on SoundPoint IP
300 and 500 phones.
reg.x.server.y.retryTimeOut
reg.x.server.y.retryMaxCount
Null or
non-negative
integer
Null
Null
Null or
non-negative
integer
reg.x.server.y.expires.lineSeize positive integer
Null
0
reg.x.server.y.lcs
0, 1
This attribute overrides the reg.x.lcs.
If set to 1, the Microsoft Live Communications
Server is supported for registration x.
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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Configuration Files
Permitted
Values
Attribute
Default
Interpretation
reg.x.ringType
1 to 22
2
The ringer to be used for calls received by
this registration. Default is the first non-silent
ringer.
reg.x.lineKeys
1 to max
1
max = the number of line keys on the phone.
max = 1 on SoundStation IP 4000,
max = 2 on IP 301, 320, 330, 430,
max = 3 on IP 501,
max = 4 on IP 550,
max = 6 on IP 600,
max = 48 on IP 601, 650 (without any
Expansion Modules attached, only 6 line keys
are available)
The number of line keys on the phone to be
associated with registration ‘x’.
reg.x.callsPerLineKey
1 to 24 OR
1 to 8
24 OR
8
For the SoundPoint IP 600, 601, and 650 the
permitted range is 1 to 24 and the default is
24.
For all other phones the permitted range is 1
to 8 and the default is 8.
This is the number of calls or conferences
which may be active or on hold per line key
associated with this registration.
Note that this overrides
call.callsPerLineKeyfor this registration.
Refer to Call Handling Configuration <call/>
reg.x.outboundProxy.address
reg.x.outboundProxy.port
dotted-decimal IP
address or host
name
Null
IP address or host name and port of a SIP
server to which the phone shall send all
requests.
1 to 65535
5060
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Permitted
Attribute
Values
Default
Interpretation
reg.x.outboundProxy.transport
DNSnaptr or
TCPpreferred or
UDPOnly or
TLS or
DNSnap
tr
If set to Null or DNSnaptr:
If reg.x.outboundProxy.address is a
hostname and reg.x.outboundProxy.port is 0
or Null, do NAPTR then SRV look-ups to try
to discover the transport, ports and servers,
as per RFC 3263. If
TCPOnly
reg.x.outboundProxy.address is an IP
address, or a port is given, then UDP is used.
If set to TCPpreferred:
TCP is the preferred transport, UDP is used if
TCP fails.
If set to UDPOnly:
Only UDP will be used.
If set to TLS:
If TLS fails, transport fails. Leave port field
empty (will default to 5061) or set to 5061.
If set to TCPOnly:
Only TCP will be used.
NOTE: TLS is not supported on SoundPoint
IP 300 and 500 phones.
reg.x.proxyRequire
string
Null
The string that needs to appear in the
“Proxy-Require” header. If Null, no
"Proxy-Require" will be sent.
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Configuration Files
Permitted
Values
Attribute
Default
Interpretation
reg.x.serverFeatureControl.cf
0, 1
0
If set to 1, server-based call forwarding is
enabled. The call server has control of call
forwarding.
If set to 0, server-based call forwarding is not
enabled. This is the old behavior.
If reg.x.serverFeatureControl.cfis not
Null, this attribute overrides the global
server-based call forwarding flag in the
sip.cfg configuration file.
reg.x.serverFeatureControl.dnd 0, 1
0
If set to 1, server-based DND is enabled. The
call server has control of DND.
If set to 0, server-based DND is not enabled.
This is the old behavior.
If reg.x.serverFeatureControl.dndis not
Null, this attribute overrides the global
server-based call forwarding flag in the
sip.cfg configuration file.
reg.x.auth.optimizedInFailover
0, 1
0
If set to 1, when failover occurs, the first new
SIP request is sent to the server that sent the
proxy authentication request.
If set to 0, when failover occurs, the first new
SIP request is sent to the server with the
highest priority in the server list.
If this parameter is Null,
voIpProt.SIP.authOptimizedInFailover
is checked.
If both parameters are set, this parameter
takes precedence.
Calls <call/>
This attribute affects the call-oriented per-phone configuration.
This attribute includes:
•
•
•
•
•
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Do Not Disturb <donotdisturb/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
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.
Automatic Off-Hook Call Placement <autoOffHook/>
An optional per-registration feature is supported which allows automatic call
placement when the phone goes off-hook.
In the following table, x is the registration number. IP 301, 320, 330, 430: x=1-2;
IP 501: x=1-3; IP 550: x=1-4; IP 600: x=1-6; IP 601: x=1-12;
IP 650: x=1-34; IP 4000: x=1.
Attribute
Permitted Values
Default
0
Interpretation
call.autoOffHook.x.enabled
call.autoOffHook.x.contact
0, 1
If set to 1, a call will be
automatically placed to
the contact specified
upon going off hook on
this registration.
ASCII encoded string containing digits
(the user part of a SIP URL) or a string
that constitutes a valid SIP URL (6416
Null
Missed Call Configuration <serverMissedCall/>
The phone supports a per-registration configuration of which events will
cause the locally displayed “missed calls” counter to be incremented.
In the following table, x is the registration number. IP 301, 320, 330, 430: x=1-2;
IP 501: x=1-3; IP 550: x=1-4; IP 600: x=1-6; IP 601: x=1-12;
IP 650: x=1-34; IP 4000: x=1 .
Permitted
Attribute
Values
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.
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Configuration Files
Missed Call Tracking <missedCallTracking/>
You can enable/disable missed call tracking on a per-line basis.
In the following table, x is the registration number. IP 301, 320, 330, 430: x=1-2;
IP 501: x=1-3; IP 550: x=1-4; IP 600: x=1-6; IP 601: x=1-12;
IP 650: x=1-34; IP 4000: x=1 .
Permitted
Attribute
Values
Default
Interpretation
call.missedCallTracking.x.enabled
0, 1
1
If set to 1 or Null, missed call tracking is
enabled.
If call.missedCallTracking.x.enabledis
set to 0, then missedCall counter is not
updated regardless of what
call.serverMissedCalls.x.enabledis set
to (and regardless of how the server is
configured). There is no Missed Call List
provided under Menu > Features of the phone.
If call.missedCallTracking.x.enabledis
set to 1 and call.serverMissedCalls.x.enabled
is set to 0, then the number of missedCall
counter is incremented regardless of how the
server is configured.
If call.missedCallTracking.x.enabledis
set to 1 and
call.serverMissedCalls.x.enabledis set
to 1, then the handling of missedCalls depends
on how the server is configured.
Call Waiting <callWaiting/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
call.callWaiting.ring
beep, ring,
silent
beep
Specifies the ring tone heard on an incoming
call when another call is active.
If set to Null, the default value is beep.
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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 301, 320, 330, 430: x=1-2;
IP 501: x=1-3; IP 550: x=1-4; IP 600: x=1-6; IP 601: x=1-12;
IP 650: x=1-34; 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 constitutes a valid SIP
URL (6416 or
Null
The forward-to contact used for
all automatic call diversion
features unless overridden by a
specific contact of a per-call
diversion feature (refer to
below).
divert.x.autoOnSpecificCaller
0, 1
1
1
If set to 1, calls may be diverted
using the Auto Divert feature of
the directory. This is a global
flag.
Note: If server-based call
forwarding is enabled, this
parameter is disabled.
divert.x.sharedDisabled
0, 1
If set to 1, all diversion features
on that line will be disabled if
the line is configured as
shared.
This attribute also includes:
•
•
•
•
Forward All <fwd/>
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
divert.fwd.x.enabled
0, 1
1
If set to 1, the user will be able to enable universal call
forwarding through the soft key menu.
Note: If server-based call forwarding is enabled, this
parameter is enabled.
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Configuration Files
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.
Note: If server-based call
forwarding is enabled, this
parameter is disabled.
divert.busy.x.timeout
divert.busy.x.contact
positive integer
60
Time in seconds to allow
altering before initiating the
diversion.
ASCII encoded string
Null
Forward-to contact for calls
forwarded due to busy status, if
Null, divert.x.contactwill be
used.
containing digits (the user part
of a SIP URL) or a string that
constitutes a valid SIP URL
(6416 or [email protected]
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
forwarded on no answer to the
contact specified.
Note: If server-based call
forwarding is enabled, this
parameter is disabled.
divert.noanswer.x.timeout
divert.noanswer.x.contact
positive integer
60
Time in seconds to allow
altering before initiating the
diversion.
ASCII encoded string
Null
Forward-to contact used for
calls forwarded due to no
answer, if Null,
divert.x.contactwill be
used.
containing digits (the user part
of a SIP URL) or a string that
constitutes a valid SIP URL
(6416 or [email protected])
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Do Not Disturb <dnd/>
The phone can automatically divert calls when Do Not Disturb (DND) is
enabled.
Attribute
Permitted Values
Default
Interpretation
divert.dnd.x.enabled
0, 1
0
If set to 1, calls will be
forwarded on DND to the
contact specified below.
Note: If server-based DND or
server-base call forwarding is
enabled, this parameter is
disabled.
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
Null
Forward-to contact used for
calls forwarded due to DND
status, if Null
divert.x.contactwill be
used.
Dial Plan <dialplan/>
Per-registration dial plan configuration is supported.
In the following tables, x is the registration number. IP 301, 320, 330, 430: x=1-2;
IP 501: x=1-3; IP 550: x=1-4; IP 600: x=1-6; IP 601: x=1-12;
IP 650: x=1-34; IP 4000: x=1.
Permitted
Attribute
Values
Default
Interpretation
dialplan.x.applyToCallListDial
0, 1
0
When present, and if
dialplan.x.digitmapis not
Null, this attribute overrides the
global dial plan defined in the
sip.cfg configuration file.
For interpretation, refer to Dial
dialplan.x.applyToDirectoryDial
0, 1
0
When present, and if
dialplan.x.digitmapis not
Null, this attribute overrides the
global dial plan defined in the
sip.cfg configuration file.
For interpretation, refer to Dial
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Configuration Files
Permitted
Values
Attribute
Default
Interpretation
dialplan.x.applyToUserDial
0, 1
1
When present, and if
dialplan.x.digitmapis not
Null, this attribute overrides the
global dial plan defined in the
sip.cfg configuration file.
For interpretation, refer to Dial
dialplan.x.applyToUserSend
dialplan.x.impossibleMatchHandling
dialplan.x.removeEndOfDial
0, 1
1
0
1
When present, and if
dialplan.x.digitmapis not
Null, this attribute overrides the
global dial plan defined in the
sip.cfg configuration file.
For interpretation, refer to Dial
0, 1 or 2
When present, and if
dialplan.x.digitmapis not
Null, this attribute overrides the
global dial plan defined in the
sip.cfg configuration file.
For interpretation, refer to Dial
0, 1
When present, and if
dialplan.x.digitmapis not
Null, this attribute overrides the
global dial plan defined in the
sip.cfg configuration file.
For interpretation, refer to Dial
This attribute also includes:
•
•
Digit Map <digitmap/>
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This configuration attribute is defined as follows:
Permitted
Values
Attribute
Default
Interpretation
dialplan.x.digitmap
A string compatible with the
digit map feature of MGCP
described in 2.1.5 of RFC
3435; string is limited to 768
bytes and 30 segments; a
comma is also allowed; a
comma is also allowed;
when reached in the digit
map, a comma will turn dial
tone back on;’+’ is allowed
as a valid digit; extension
letter ‘R’ is used as defined
above.
Null
When present, this attribute
overrides the global dial plan
defined in the sip.cfg
configuration file.
For more information, refer to
Digit Map <digitmap/> on page
A-17.
dialplan.x.digitmap.timeOut
string of positive integers
separated by ‘|’
Null
When present, and if
dialplan.x.digitmapis not
Null, this attribute overrides the
global dial plan defined in the
sip.cfg configuration file.
For more information, refer to
Digit Map <digitmap/> on page
A-17.
Routing <routing/>
This attribute allows specific routing paths for outgoing SIP calls to be
configured independent of other ‘default’ configuration.
This attribute includes:
•
•
Server <server/>
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
dialplan.x.routing.server.y.add dotted-decimal IP address
ress 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.routing.server.y.port 1 to 65535
5060
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Configuration Files
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.
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
dialplan.x.routing.emergency.
y.value
Comma separated list of
entries or single entry
representing a or a
Null
This represents the URLs
that should be watched for
emergency routing.
Example:
“15,17,18”, “911”,
“sos”.
combination of SIP URL.
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.emergency.
y.server.z
positive integer
Null
Index representing the
server defined in Server
will be used for emergency
routing.
Messaging <msg/>
Message-waiting indication is supported on a per-registration basis.
This configuration attribute is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
msg.bypassInstantMessage
0, 1
0
If set to 1, the display offering a choice of
“Message Center” and “Instant Messages” will
be bypassed when pressing the Messages key.
The phone will act as if “Message Center” was
accessible from the Main Menu.
This attribute also includes:
•
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Message Waiting Indicator <mwi/>
In the following table, x is the registration number. IP 301, 320, 330, 430: x=1-2;
IP 501: x=1-3; IP 550: x=1-4; IP 600: x=1-6; IP 601: x=1-12;
IP 650: x=1-34; IP 4000: x=1.
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
msg.mwi.x.subscribe
ASCII encoded string containing
digits (the user part of a SIP
URL) or a string that constitutes
a valid SIP URL (6416 or
Null
If non-Null, the phone will send
a SUBSCRIBE request to this
contact after boot-up.
msg.mwi.x.callBackMo
de
contact or
registration or
disabled
“registration”
Configures message retrieval
and notification for the line.
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 contact registered (the
phone will call itself).
If set to “disabled”, message
retrieval and message
notification are disabled.
msg.mwi.x.callBack
ASCII encoded string containing
digits (the user part of a SIP
URL) or a string that constitutes
a valid SIP URL (6416 or
Null
Contact to call when retrieving
messages for this registration.
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.
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Configuration Files
This configuration attribute is defined as follows:
Permitted
Values
Attribute
Default
Interpretation
nat.ip
dotted-decima Null
l IP address
IP address to advertise within SIP signaling - should
match the external IP address used by the NAT device.
nat.signalPort
1024 to 65535 Null
If non-Null, this port will be used by the phone for SIP
signaling, overriding the value set for
voIpProt.local.signalPortin sip.cfg.
nat.mediaPortStart
1024 to 65535 Null
If non-Null, this attribute will be used to set the initially
allocated RTP port, overriding the value set for
tcpIpApp.port.rtp.mediaPortRangeStartin sip.cfg.
nat.keepalive.interval
0 to 3600
Null
If non-Null (or 0), the keepalive interval in seconds. This
parameter is used to set the interval at which phones will
send a keep-alive packet to the gateway/NAT device to
keep the communication port open so that NAT can
continue to function as setup initially.
The Microsoft Live Communications Server 2005
keepalive feature will override this interval. If you want to
deploy phones behind a NAT and connect them to Live
Communications Server, the keepalive interval received
from the Live Communications Server must be short
enough to keep the NAT port open. Once the TCP
connection is closed, the phones stop sending keep-alive
packets.
Attendant <attendant/>
These attributes are available on SoundPoint IP 600 and 601 phones (with an
attached Expansion Module) only.
Note
The Busy Lamp Field (BLF) / attendant console feature enhances support for
a phone-based attendant console.
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Administrator’s Guide SoundPoint IP / SoundStation IP
This configuration attribute is defined as follows:
Permitted
Values
Attribute
Default
Interpretation
attendant.uri
string
Null
For attendant console / busy lamp field (BLF) feature.
This specifies the list SIP URI on the server. If this is just
a user part, the URI is constructed with the server host
name/IP.
attendant.reg
positive
integer
1
For attendant console / BLF feature. This is the index of
the registration which will be used to send a SUBSCRIBE
to the list SIP URI specified in attendant.uri. For example,
attendant.reg = 2 means the second registration will be
used.
Roaming Buddies <roaming_buddies/>
This attribute is used in conjunction with Microsoft Live Communications
Server 2005 only.
Note
This configuration attribute is defined as follows:
Permitted
Attribute
roaming_buddies.reg
Values
Default
Interpretation
positive
integer
Null
Specifies the line/registration number which has roaming
buddies support enabled. If Null, roaming buddies is
disabled. If value < 1, then value is replaced with 1.
Warning: This parameter must be enabled
(value > 0) if the call server is Microsoft Live
Communications Server 2005.
Roaming Privacy <roaming_privacy/>
This attribute is used in conjunction with Microsoft Live Communications
Server 2005 only.
Note
This configuration attribute is defined as follows:
Permitted
Attribute
roaming_privacy.reg
Values
Default
Interpretation
positive
integer
Null
Specifies the line/registration number which has roaming
privacy support enabled. If Null, roaming privacy is
disabled. If value < 1, then value is replaced with 1.
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Configuration Files
Flash Parameter Configuration
Any field in the bootROM setup menu and the application SIP Configuration
menu can be set through a configuration file.
A DHCP server can be configured to point the phones to a boot server that has
the required configuration files. The new settings will be downloaded by the
phones and used to configure them. This removes the need for manual
interaction with phones to configure basic settings. This is especially useful for
initial installation of multiple phones.
These device settings are detected when the application starts. If the new
settings would normally cause a reboot if they were changed in the application
Network Configuration menu, then they will cause a reboot when the
application starts.
The parameters for this feature should be put in separate configuration files to
simplify maintenance. Do not add them to existing configuration files (such as
sip.cfg). One new configuration file will be required for parameters that should
apply to all phones, and individual configuration files will be required for
phone-specific parameters such as SIP registration information.
Caution
The global device.setparameter must be enabled when the initial
installation is done, and then it should be disabled. This prevents subsequent
reboots by individual phones triggering a reset of parameters on the phone
that may have been tweaked since the initial installation.
This feature is very powerful and should be used with caution. For example, an
incorrect setting could set the IP Address of multiple phones to the same value.
Caution
Note that some parameters may be ignored, for example if DHCP is enabled it will
still override the value set with device.net.ipAddress
.
Individual parameters are checked to see whether they are in range, however, the
interaction between parameters is not checked. If a parameter is out of range, an
error message will appear in the log file and parameter will not be used.
Incorrect configuration could cause phones to get into a reboot loop. For example,
server A has a configuration file that specifies that server B should be used, which
has a configuration file that specifies that server A should be used.
Polycom recommends that you test the new configuration files on two phones
before initializing all phones. This should detect any errors including IP address
conflicts.
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This flash attributes are defined as follows:
Name
Possible Values
Description
device.set
0 or 1
default = 0
If set to 0, do not use any device.xxx.yyyfields to
set any parameters. Set this to 0 after the initial
installation.
If set to 1, use the device.xxx.yyyfields that have
device.xxx.yyy.set= 1. Set this to 1 for the initial
installation only.
device.xxx.yyy.set
0 or 1
If set to 0, do not use the device.xxx.yyyvalue.
default = 0
If set to 1, use the device.xxx.yyyvalue.
For example, if device.net.ipAddress.set= 1,
then use the contents of the device.net.ipAddress
field.
device.net.ipAddress
device.net.subnetMask
device.net.IPgateway
dotted-decimal IP address
dotted-decimal IP address
dotted-decimal IP address
Phone's IP address.
Note: This field is not used when DHCP client is
enabled.
Phone's subnet mask.
Note: This field is not used when DHCP client is
enabled.
Phone's default router / IP gateway.
Note: This field is not used when DHCP client is
enabled.
device.net.vlanId
Null, 0 to 4094
0 or 1
Phone’s 802.1Q VLAN identifier.
Note: Null = no VLAN tagging
device.net.cdpEnabled
device.dhcp.enabled
device.dhcp.offerTimeout
If set to 1, the phone will attempt to determine its
VLAN ID through the CDP.
0 or 1
For description, refer to DHCP or Manual TCP/IP
1 to 600
Number of seconds the phone waits for secondary
DHCP Offer messages before selecting an offer.
device.dhcp.bootSrvUseOp 0 to 3
t
device.dhcp.bootSrvOpt
128 to 254 (Cannot be the
same as VLAN ID Option)
device.dhcp.bootSrvOptTy
pe
0 or 1
device.dhcp.dhcpVlanDisc
UseOpt
0 to 2
device.dhcp.dhcpVlanDisc
Opt
128 to 254 (Cannot be the
same as Boot Server
Option)
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Configuration Files
Name
Possible Values
any string
0 to 4
Description
device.prov.serverName
device.prov.serverType
device.prov.user
any string
any string
0 or 1
device.prov.password
device.prov.appProvType
device.prov.appProvString
any string
device.prov.redunAttemptLi 10, Null
mit
device.prov.redunInterAtte
mptDelay
300, Null
device.sntp.serverName
any string
Can be dotted-decimal IP address or domain name
string. SNTP server from which the phone will obtain
the current time
device.sntp.gmtOffset
device.dns.serverAddress
device.dns.altSrvAddress
device.dns.domain
-43200 to 46800
GMT offset in seconds, corresponding to -12 to +13
hours.
dotted-decimal IP address
dotted-decimal IP address
Primary server to which the phone directs Domain
Name System queries.
Secondary server to which the phone directs Domain
Name System queries.
any string
any string
The phone’s DNS domain.
device.auth.localAdminPas
sword
The phone’s local administrator password.
device.auth.localUserPass
word
any string
any string
any string
any string
The phone user’s local password.
device.auth.regUserx
The SIP registration user name for registration x
where x = 1 to 12.
device.auth.regPasswordx
The SIP registration password for registration x
where x = 1 to 12.
device.sec.configEncryptio
n.key
Configuration encryption key that is used for
encryption of configuration files.
device.syslog.serverName
dotted-decimal IP address
OR
domain name string
The syslog server IP address or host name.
The default value is NULL.
device.syslog.transport
None=0,
UDP=1,
TCP=2,
TLS=3
The protocol that the phone will use to write to the
syslog server.
If set to “None”, transmission is turned off, but the
server address is preserved.
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Administrator’s Guide SoundPoint IP / SoundStation IP
Name
Possible Values
Description
device.syslog.facility
0 to 23
A description of what generated the log message.
For more information, refer to section 4.1.1 of RFC
3165.
The default value is 16, which maps to “local 0”.
device.syslog.renderLevel
1 to 6
Specifies the lowest class of event that will be
rendered to syslog. It is based on
log.render.level and can be a lower value.
Refer to Basic Logging <level/><change/> and
device.syslog.prependMac
device.em.power
Enabled, Disabled
Enabled, Disabled, Null
Enabled, Disabled
Enabled, Disabled
If enabled, the phone’s MAC address is prepended
to the log message sent to the syslog server.
device.net.etherVlanFilter
device.net.etherStormFilter
Refer to the VLAN Filtering parameter in Ethernet
Refer to the Storm Filtering parameter in Ethernet
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B
Session Initiation Protocol (SIP)
This chapter provides a description of the basic Session Initiation Protocol
(SIP) and the protocol extensions that are supported by the current SIP
application. To find the applicable Request For Comments (RFC) document,
This chapter contains information on:
•
Basic Protocols—All the basic calling functionality described in the SIP
specification is supported. Transfer is included in the basic SIP support.
•
Protocol Extensions—Extensions add features to SIP that are applicable to
a range of applications, including reliable 1xx responses and session
timers.
For information on supported RFC’s and Internet drafts, refer to the following
This chapter also describes:
•
•
•
•
•
•
•
•
•
•
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Administrator’s Guide SoundPoint IP / SoundStation IP
RFC and Internet Draft Support
The following RFC’s and Internet drafts are supported:
•
•
•
•
•
•
RFC 1321—The MD5 Message-Digest Algorithm
RFC 2327—SDP: Session Description Protocol
RFC 2387—The MIME Multipart / Related Content-type
RFC 2976—The SIP INFO Method
RFC 3261—SIP: Session Initiation Protocol (replacement for RFC 2543)
RFC 3262—Reliability of Provisional Responses in the Session Initiation
Protocol (SIP)
•
•
RFC 3263—Session Initiation Protocol (SIP): Locating SIP Servers
RFC 3264—An Offer / Answer Model with the Session Description
Protocol (SDP)
•
•
•
•
•
•
•
•
RFC 3265—Session Initiation Protocol (SIP) - Specific Event Notification
RFC 3311—The Session Initiation Protocol (SIP) UPDATE Method
RFC 3325—SIP Asserted Identity
RFC 3515—The Session Initiation Protocol (SIP) Refer Method
RFC 3555 — MIME Type of RTP Payload Formats
RFC 3665—Session Initiation Protocol (SIP) Basic Call Flow Examples
draft-ietf-sip-cc-transfer-05.txt—SIP Call Control - Transfer
RFC 3725—Best Current Practices for Third Party Call Control (3pcc) in
the Session Initiation Protocol (SIP)
•
RFC 3842—A Message Summary and Message Waiting Indication Event
Package for the Session Initiation Protocol (SIP)
•
•
•
•
RFC 3856—A Presence Event Package for Session Initiation Protocol (SIP)
RFC 3891—The Session Initiation Protocol (SIP) “Replaces” Header
RFC 3892—The Session Initiation Protocol (SIP) Referred-By Mechanism
RFC 3959—The Early Session Disposition Type for the Session Initiation
Protocol (SIP)
•
•
RFC 3960—Early Media and Ringing Tone Generation in the Session
Initiation Protocol (SIP)
RFC 3968—The Internet Assigned Number Authority (IANA) Header
Field Parameter Registry for the Session Initiation Protocol (SIP)
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Session Initiation Protocol (SIP)
•
RFC 3969—The Internet Assigned Number Authority (IANA) Uniform
Resource Identifier (URI) Parameter Registry for the Session Initiation
Protocol (SIP)
•
•
RFC 4028—Session Timers in the Session Initiation Protocol (SIP)
RFC 4235—An INVITE-Initiated Dialog Event Package for the Session
Initiation Protocol (SIP)
•
RFC 4662—Session Initiation Protocol (SIP) Event Notification Extension
for Resource Lists
•
•
draft-levy-sip-diversion-04.txt—Diversion Indication in SIP
draft-anil-sipping-bla-02.txt—Implementing Bridged Line Appearances
(BLA) Using Session Initiation Protocol (SIP)
•
draft-ietf-sip-privacy-04.txt—SIP Extensions for Network-Asserted Caller
Identity and Privacy within Trusted Networks
•
•
draft-levy-sip-diversion-06.txt—Diversion Indication in SIP
draft-ietf-sipping-cc-conferencing-03.txt—SIP Call Control - Conferencing
for User Agents
•
draft-ietf-sip-connect-reuse-04.txt—Connection Reuse in the Session
Initiation Protocol (SIP)
Request Support
The following SIP request messages are supported:
Method
REGISTER
INVITE
Supported
Yes
Notes
Yes
ACK
Yes
CANCEL
BYE
Yes
Yes
OPTIONS
SUBSCRIBE
NOTIFY
REFER
PRACK
Yes
Yes
Yes
Yes
Yes
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Administrator’s Guide SoundPoint IP / SoundStation IP
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.
Header Support
The following SIP request headers are supported:
In the following table, a “Yes” in the Supported column means the header is sent
and properly parsed.
Note
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
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Session Initiation Protocol (SIP)
Header
Supported
Yes
No
Notes
Diversion
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
Yes
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Administrator’s Guide SoundPoint IP / SoundStation IP
Header
Supported
No
Notes
Subject
Subscription-State
Yes
Yes
No
Supported
Timestamp
To
Yes
No
Unsupported
User-Agent
Via
Yes
Yes
No
Warning
WWW-Authenticate
Yes
Response Support
The following SIP responses are supported:
In the following table, a “Yes” in the Supported column means the header is sent
and properly parsed. The phone may not actually generate the response.
Note
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
2xx Responses - Success
Response
200 OK
Supported
Yes
Notes
202 Accepted
Yes
In REFER transfer.
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Session Initiation Protocol (SIP)
3xx Responses - Redirection
Response
Supported
Notes
300 Multiple Choices
301 Moved Permanently
302 Moved Temporarily
305 Use Proxy
Yes
Yes
Yes
No
380 Alternative Service
No
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.
Note
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
480 Temporarily Unavailable
No
No
Yes
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Administrator’s Guide SoundPoint IP / SoundStation IP
Response
Supported
Yes
Yes
No
Notes
481 Call/Transaction Does Not Exist
482 Loop Detected
483 Too Many Hops
484 Address Incomplete
485 Ambiguous
Yes
No
486 Busy Here
Yes
Yes
Yes
No
487 Request Terminated
488 Not Acceptable Here
491 Request Pending
493 Undecipherable
No
5xx Responses - Server Failure
Response
Supported
Notes
500 Server Internal Error
501 Not Implemented
502 Bad Gateway
Yes
Yes
No
503 Service Unavailable
504 Server Time-out
505 Version Not Supported
513 Message Too Large
No
No
No
No
6xx Responses - Global Failure
Response
Supported
Notes
600 Busy Everywhere
603 Decline
No
Yes
No
No
604 Does Not Exist Anywhere
606 Not Acceptable
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Session Initiation Protocol (SIP)
Hold Implementation
The phone supports both currently accepted means of signaling hold.
The first method, no longer recommended due in part to the RTCP problems
associated with it, is to set the “c” destination addresses for the media streams
in the SDP to zero, for example, c=0.0.0.0.
The second, and preferred, method is to signal the media directions with the
“a” SDP media attributes sendonly, recvonly, inactive, or sendrecv. The hold
point.
Even if the phone is set to use c=0.0.0.0, it will not do so if it gets any sendrecv,
sendonly, or inactive from the server. These flags will cause it to revert to the other
hold method.
Note
Reliability of Provisional Responses
The phone fully supports RFC 3262 - Reliability of Provisional Responses.
Transfer
The phone supports transfer using the REFER method specified in
draft-ietf-sip-cc-transfer-05 and RFC 3515.
Third Party Call Control
The phone supports the delayed media negotiations (INVITE without SDP)
associated with third party call control applications.
SIP for Instant Messaging and Presence Leveraging Extensions
The phone is compatible with the Presence and Instant Messaging features of
Microsoft Windows Messenger 5.1. In a future release, support for the
Presence and Instant Message recommendations in the SIP Instant Messaging
and Presence Leveraging Extensions (SIMPLE) proposals will be provided by
the following Internet drafts or their successors:
•
•
•
•
•
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
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Administrator’s Guide SoundPoint IP / SoundStation IP
Shared Call Appearance Signaling
A shared line is an address of record managed by a call server. The server
allows multiple end points to register locations against the address of record.
The phone supports shared call appearances (SCA) using the
SUBSCRIBE-NOTIFY method in the “SIP Specific Event Notification”
framework (RFC 3265). The events used are:
•
•
“call-info” for call appearance state notification
“line-seize for the phone to ask to seize the line
Bridged Line Appearance Signaling
A bridged line is an address of record managed by a server. The server allows
multiple end points to register locations against the address of record.
The phone supports bridged line appearances (BLA) using the
SUBSCRIBE-NOTIFY method in the “SIP Specific Event Notification”
framework (RFC 3265). The events used are:
•
“dialog” for bridged line appearance subscribe and notify
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C
Miscellaneous Administrative Tasks
This appendix provides information required by varied aspects of the Session
Initiation Protocol (SIP) application. This includes:
•
•
•
•
•
•
•
•
Trusted Certificate Authority List
The following certificate authorities are trusted by the phone by default:
•
•
•
•
•
•
•
•
•
ABAecom (sub., Am. Bankers Assn.) Root CA
ANX Network CA by DST
American Express CA
American Express Global CA
BelSign Object Publishing CA
BelSign Secure Server CA
Deutsche Telekom AG Root CA
Digital Signature Trust Co. Global CA 1
Digital Signature Trust Co. Global CA 2
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Administrator’s Guide SoundPoint IP / SoundStation IP
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
Digital Signature Trust Co. Global CA 3
Digital Signature Trust Co. Global CA 4
Entrust Worldwide by DST
Entrust.net Premium 2048 Secure Server CA
Entrust.net Secure Personal CA
Entrust.net Secure Server CA
Equifax Premium CA
Equifax Secure CA
GTE CyberTrust Global Root
GTE CyberTrust Japan Root CA
GTE CyberTrust Japan Secure Server CA
GTE CyberTrust Root 2
GTE CyberTrust Root 3
GTE CyberTrust Root 4
GTE CyberTrust Root 5
GTE CyberTrust Root CA
GlobalSign Partners CA
GlobalSign Primary Class 1 CA
GlobalSign Primary Class 2 CA
GlobalSign Primary Class 3 CA
GlobalSign Root CA
National Retail Federation by DST
TC TrustCenter, Germany, Class 1 CA
TC TrustCenter, Germany, Class 2 CA
TC TrustCenter, Germany, Class 3 CA
TC TrustCenter, Germany, Class 4 CA
Thawte Personal Basic CA
Thawte Personal Freemail CA
Thawte Personal Premium CA
Thawte Premium Server CA
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Miscellaneous Administrative Tasks
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
Thawte Server CA
Thawte Universal CA Root
UPS Document Exchange by DST
ValiCert Class 1 VA
ValiCert Class 2 VA
ValiCert Class 3 VA
VeriSign Class 4 Primary CA
Verisign Class 1 Public Primary Certification Authority
Verisign Class 1 Public Primary Certification Authority - G2
Verisign Class 1 Public Primary Certification Authority - G3
Verisign Class 2 Public Primary Certification Authority
Verisign Class 2 Public Primary Certification Authority - G2
Verisign Class 2 Public Primary Certification Authority - G3
Verisign Class 3 Public Primary Certification Authority
Verisign Class 3 Public Primary Certification Authority - G2
Verisign Class 3 Public Primary Certification Authority - G3
Verisign Class 4 Public Primary Certification Authority - G2
Verisign Class 4 Public Primary Certification Authority - G3
Verisign/RSA Commercial CA
Verisign/RSA Secure Server CA
Encrypting Configuration Files
The phone can recognize encrypted files, which it downloads from the boot
server and it can encrypt files before uploading them to the boot server. There
must be an encryption key on the phone to perform these operations.
Configuration files (excluding the master configuration file), contact
directories, and configuration override files can be encrypted.
A separate SDK, with a readme file, is provided to facilitate key generation and
configuration file encryption and decrypt on a UNIX or Linux server. The
utility is distributed as source code that runs under the UNIX operating
system. For more information, contact Polycom Technical Support.
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Administrator’s Guide SoundPoint IP / SoundStation IP
A key is generated by the utility and must be downloaded to the phone so that
it can decrypt the files that were encrypted on the server. The
device.sec.configEncryption.keyconfiguration file parameter is used to
set the key on the phone. The utility generates a random key and the
encryption is Advanced Encryption Standard (AES) 128 in Cipher Block
Chaining (CBC) mode. An example key would look like this:
Crypt=1;KeyDesc=companyNameKey1;Key=06a9214036b8a15b512e03d534120006;
If the phone doesn't have a key, it must be downloaded to the phone in plain
text (a potential security hole if not using HTTPS). If the phone already has a
key, a new key can be downloaded to the phone encrypted using the old key
phones from the factory will have a key pre-loaded in them. This key will be
changed at regular intervals to enhance security
It is recommended that all keys have unique descriptive strings in order to
allow simple identification of which key was used to encrypt a file. This makes
boot server management easier.
After encrypting a configuration file, it is useful to rename the file to avoid
confusing it with the original version, for example rename sip.cfg to sip.enc.
However, the directory and override filenames cannot be changed in this
manner.
You can check whether an encrypted file is the same as an unencrypted file by:
1. Run the configFileEncrypt utility on the unencrypted file with the "-d"
option. This shows the "digest" field.
2. Look at the encrypted file using WordPad and check the first line that
shows a "Digest=…." field. If the two fields are the same, then the
encrypted and unencrypted file are the same.
If a phone downloads an encrypted file that it cannot decrypt, the action is logged,
an error message displays, and the phone reboots. The phone will continue to do
this until the boot server provides an encrypted file that can be read, an
unencrypted file, or the file is removed from the master configuration file list.
Note
Note
The SoundPoint IP 300 and 500 phones will always fail at decrypting files. These
phones will recognize that a file is encrypted, but cannot decrypt it and will display
an error. This information is logged. Encrypted configuration files can only be
decrypted on the SoundPoint IP 301, 320, 330, 430, 501,550, 600, 601, and 650
and the SoundStation IP 4000 phones.
The master configuration file cannot be encrypted on the boot server. This file is
downloaded by the bootROM that does not recognize encrypted files. For more
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The following configuration file changes are required to modify this feature:
Central
Configuration File: sip.cfg
Specify the phone-specific contact directory and the
phone-specific configuration override file.
(boot server)
For more information, refer to Encryption
Configuration file:
Change the encryption key.
<device>.cfg
For more information, refer to Flash Parameter
Changing the Key on the Phone
For security purposes, it may be desirable to change the key on the phones and
the server from time to time.
To change a key:
1. Put the new key into a configuration file that is in the list of files
downloaded by the phone (specified in 000000000000.cfg or <Ethernet
address>.cfg).
Use the device.sec.configEncryption.keyparameter to specify the
new key.
2. Manually reboot the phone so that it will download the new key. The
phone will automatically reboot a second time to use the new key.
At this point, the phone expects all encrypted configuration files on the
boot server to use the new key and it will continue to reboot until this is
the case. The files on the server must be updated to the new key or they
must be made available in unencrypted format. Updating to the new key
requires decrypting the file with the old key, then encrypting it with the
new key.
Note that configuration files, contact directory files and configuration
override files may all need to be updated if they were already encrypted.
In the case of configuration override files, they can be deleted from the
boot server so that the phone will replace them when it successfully boots.
Adding a Background Logo
This section provides instructions on how to add a background logo to all
SoundPoint IP phones in your organization. You must be running at least
BootROM 2.x.x and SIP 1.x.x. One bitmap file is required for each model, but
SoundPoint IP 301 phones do not support bitmap logos.
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Administrator’s Guide SoundPoint IP / SoundStation IP
Model
Width
n/a
Height
n/a
Color Depth
n/a
IP 301
IP 320/330
IP 430
225
94
75
monochrome
monochrome
23
IP 501
114
51
4-bit grayscale or
monochrome
IP 550/600/601
IP 650
209
209
150
109
109
33
4-bit grayscale or
monochrome
16-bit grayscale or
monochrome
IP 4000
monochrome
Logos smaller than described in the table above are acceptable, but larger
logos may be truncated or interfere with other areas of the user interface.
The SoundPoint IP 500/501/550/600/601 phones only support the four colors
black, dark gray, light gray, and white. Any other colors will be approximated.
RGB Values
Color
RGB Values (Decimal)
0,0,0
(Hexadecimal)
Black
00,00,00
Dark Gray
Light Gray
White
96,96,96
60,60,60
160,160,160
255,255,255
A0,A0,A0
FF,FF,FF
The SoundPoint IP 650 phones support a 16-bit grayscale, which is a smooth
gradient from black (0, 0, 0) to white (FF, FF, FF).
The SoundStation IP 4000 phone only supports black and white. Any other
colors will be rendered as either black or white.
Configuration File Changes
In the <bitmaps> section of sip.cfg, find the end of each model's bitmap list and
add your bitmap to the end; do not include the .bmp extension:
<bitmaps>
<IP_300 … />
<IP_330 … bitmap.IP_330.66.name="logo-330" />
<IP_500 … bitmap.IP_500.66.name="logo-500" />
<IP_600 … bitmap.IP_600.70.name="logo-600" />
<IP_4000 … bitmap.IP_4000.70.name="logo-4000" />
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</bitmaps>
Next, enable the idle display feature and modify the idle display "animation"
for each model to point to your bitmap (again without the .bmp extension):
<indicators ind.idleDisplay.enabled="1">
<Animations>
<IP_300>
…
</IP_300>
<IP_330>
…
<IDLE_DISPLAY ind.anim.IP_3300.38.frame.1.bitmap="logo-330"
ind.anim.IP_330.38.frame.1.duration="0"/>
…
</IP_330>
<IP_500>
…
<IDLE_DISPLAY ind.anim.IP_500.38.frame.1.bitmap="logo-500"
ind.anim.IP_500.38.frame.1.duration="0"/>
…
</IP_500>
<IP_600>
…
<IDLE_DISPLAY ind.anim.IP_600.38.frame.1.bitmap="logo-600"
ind.anim.IP_600.38.frame.1.duration="0"/>
…
</IP_600>
<IP_4000>
…
<IDLE_DISPLAY ind.anim.IP_4000.38.frame.1.bitmap="logo-4000"
ind.anim.IP_4000.38.frame.1.duration="0"/>
…
</IP_4000>
</Animations>
…
</indicators>
BootROM/SIP Application Dependencies
Not withstanding the hardware backward compatibility mandate, there have
been times throughout the life of the SoundPoint IP / SoundStation IP phones
where certain dependencies on specific bootROM and application versions
have been necessitated.
This table summarizes some the major dependences that you are likely to
encounter:
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Administrator’s Guide SoundPoint IP / SoundStation IP
Model
BootROM
SIP Application
IP 301/501
2.6.1 or later
1.4.2, 1.5.4, 1.6.1 or
later
IP 320/330
IP 430
3.2.3 B or later
3.1.3 C or later
3.2.2 B or later
2.0 or later
2.1.1 or later
1.6.6 or later
2.1 or later
IP 550
IP 600
1.0 or later
IP 601/EM
IP 650/EM
IP 4000
3.1 or later
1.6 or later
3.2.2 B or later
2.6 or later
2.0.3 B or later
1.4 or later
Migration Dependencies
In addition to the bootROM and application dependencies, there are certain
restrictions with regard to upgrading or downgrading from one bootROM
release to another bootROM release. These restrictions are typically caused by
the addition of features that change the way bootROM provisioning is done,
so the older version become incompatible.
There is always a way to move forward with bootROM releases, although it
may be a two or three step procedure sometimes, but there are cases where it
is impossible to move backward. Make special note of these cases before
upgrading.
Note that:
•
•
•
•
1.x cannot be upgraded to any 2.x automatically
2.0 and 2.1 can not upgrade past 2.4
Only 2.6 can upgrade to 3.0
3.0 cannot be downgraded
For example, a two step upgrade would be necessary from bootROM 2.1 to
bootROM 2.5. A direct upgrade is not supported, but upgrading to bootROM
2.2 first, then upgrading to 2.5 will work.
Downgrade restrictions are limited to major releases. Going from 2.x to 1.x and
from 3.x to 2.x are both impossible in the field.
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Miscellaneous Administrative Tasks
Multiple Key Combinations
On SoundPoint IP and SoundStation IP phones, certain multiple key
combinations can be used to reboot the phone and restore factory defaults.
For other methods for resetting and rebooting your SoundPoint IP or
SoundStation IP phones, refer to “Quick Tip 18298: Resetting and Rebooting
Rebooting the Phone
For the key combination, press and hold certain key combinations (depending
on the phone model) simultaneously until a confirmation tone is heard or for
about three seconds :
•
•
•
•
•
IP 301: Volume-, Volume+, Hold, and Do Not Disturb
IP 320 and 330: Volume-, Volume+, Hold, and Hands-free
IP 430 and 501: Volume-, Volume+, Hold, and Messages
IP 550, 600, 601, and 650: Volume-, Volume+, Mute, and Messages
IP 4000: *, #, Volume+, and Select
Restoring Factory Defaults
For the key combination, press and hold certain key combinations (depending
on the phone model) simultaneously during the countdown process in the
bootROM until the password prompt appears:
•
•
•
IP 301, 501, 550, 600, 601, and 650: 4, 6, 8 and * dial pad keys
IP 320, 330, and 430: 1, 3, 5, and 7 dial pad keys
IP 4000: 6, 8 and * dial pad keys
Enter the administrator password to initiate the reset. Resetting to factory
defaults will also reset the administrator password (factory default password
is 456).
Uploading Log Files
For the key combination, press and hold certain key combinations (depending
on the phone model) simultaneously until a confirmation tone is heard or for
about three seconds:
•
•
•
•
IP 301: The two Line keys and the Up and Down arrow keys
IP 320 and 330: Menu, Dial, and the two Line keys
IP 430, 501, 550, 600, 601, 650: Up, Down, Left, and Right arrow keys
IP 4000: Menu, Exit, Off-hook/Hands-free, Redial
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Administrator’s Guide SoundPoint IP / SoundStation IP
Default Feature Key Layouts
The following figures and table show the default SIP key layouts for
SoundPoint IP 301, 320, 330, 430, 501, 550, 600, 601, and 650 and SoundStation
IP 4000 models.
SoundPoint IP 301
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 320/330
31
13
7
14
15
Menu
Dial
Line 1
Line 2
32
34
8
33
9
10
16
ABC
DEF
1 6 2 1 325
Hold
GHI
JKL
MNO
19
20
21
4
5 2 626
5
PQRS
TUV
WXYZ
7 4 8 3 927
OPER
30
028
29
24
22
23
Key ID
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SoundPoint IP 430
SoundPoint IP 501
1
35
34
Sel
40
2
4
3
5
39
33
32
31
30
29
37
36
Del
38 Directories
Services
Call Lists
Conference
Transfer
Redial
6
Menu
28
25
27
26
7
8
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
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Administrator’s Guide SoundPoint IP / SoundStation IP
SoundPoint IP 550/600/601/650
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
15
OPER
Hold
0
40
13
11
12
Key ID
The SoundPoint IP 550 has only the top four lines keys.
Note
SoundStation IP 4000
6
25
26
12
18
27
28
29
4
1
7
2
8
3
5
10
9
16
22
13
19
14
20
15
21
IP 320 &
330
Function
IP 550, 600,
601, & 650
Function
Key IP 301
IP 430
Function
IP 501
Function
IP 4000
Function
ID
Function
1
Line1
Line2
n/a
Dialpad2
Dialpad5
Dialpad8
Dialpad7
Dialpad4
Line1
ArrowUp
ArrowLeft
Select
ArrowUp
ArrowLeft
ArrowDown
ArrowRight
Select
Dialpad1
Dialpad2
Dialpad3
VolUp
2
Line2
3
n/a
4
n/a
ArrowUp
Hold
ArrowRight
5
Hold
ArrowDown
Handsfree
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IP 320 &
330
Function
IP 550, 600,
601, & 650
Function
Key IP 301
IP 430
Function
IP 501
Function
IP 4000
Function
ID
Function
6
n/a
Dialpad1
SoftKey3
Line1
n/a
Delete
Delete
ArrowUp
Dialpad4
Dialpad5
Dialpad6
VolDown
n/a
7
Redial
Redial
Menu
Menu
8
VolUp
VolUp
Messages
DoNotDisturb
Hold
Messages
DoNotDisturb
MicMute
VolUp
9
VolDown
DialpadStar
Dialpad0
DialpadPound
Dialpad9
Dialpad8
Dialpad7
Dialpad4
Dialpad5
Dialpad6
Dialpad3
Dialpad2
Dialpad1
n/a
ArrowRight
Line2
VolDown
DialpadStar
Dialpad0
DialpadPound
Dialpad9
Dialpad8
Dialpad7
Dialpad4
Dialpad5
Dialpad6
Dialpad3
Dialpad2
Dialpad1
ArrowRight
Messages
n/a
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
n/a
VolUp
n/a
VolDown
DialpadPound
Dialpad0
DialpadStar
Dialpad9
Dialpad8
Dialpad7
Dialpad4
Dialpad5
Dialpad6
Dialpad3
Dialpad2
Dialpad1
SoftKey4
SoftKey3
SoftKey2
SoftKey1
Conference
CallHistory
Services
Directories
Line3
VolDown
DialpadPound
Dialpad0
DialpadStar
Dialpad9
Dialpad8
Dialpad7
Dialpad4
Dialpad5
Dialpad6
Dialpad3
Dialpad2
Dialpad1
SoftKey4
SoftKey3
SoftKey2
SoftKey1
Services
Directories
Line6
Select
Dialpad7
Dialpad8
Dialpad9
MicMute
n/a
SoftKey2
ArrowUp
Select
ArrowDown
n/a
n/a
ArrowDown
DialpadStar
Dialpad0
DialpadPound
Redial
n/a
Hold
Headset
Handsfree
DialpadPound
VolUp
DoNotDisturb
n/a
VolDown
Dialpad3
Dialpad6
Dialpad9
Dialpad0
DialpadStar
MicMute
SoftKey1
Dial
n/a
SoftKey3
MicMute
SoftKey2
SoftKey1
ArrowDown
n/a
SoftKey4
Headset
SoftKey2
SoftKey1
ArrowDown
Select
Menu
Exit
SoftKey1
SoftKey2
SoftKey3
n/a
ArrowUp
Menu
ArrowLeft
Menu
n/a
Conference
Line2
n/a
n/a
ArrowLeft
Menu
MicMute
SoftKey3
Handsfree
n/a
n/a
n/a
Line2
Line1
n/a
Headset
n/a
n/a
Line1
Line3
n/a
n/a
Redial
Redial
n/a
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IP 320 &
330
Function
IP 550, 600,
601, & 650
Function
Key IP 301
IP 430
Function
IP 501
Function
IP 4000
ID
37
38
39
40
41
42
Function
Function
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
n/a
n/a
n/a
n/a
n/a
Transfer
Headset
MicMute
Handsfree
n/a
Transfer
Headset
Handsfree
Hold
n/a
n/a
n/a
n/a
n/a
n/a
Line4
n/a
Line5
The SoundPoint IP 550 has only the top four lines keys. Key IDs 31 and 42 are not
used on the SoundPoint IP 550 platform.
Note
Assigning a VLAN ID Using DHCP
To assign a VLAN ID to a phone using DHCP:
>> In the DHCP menu of the Main setup menu, set VLAN Discovery to
“Fixed” or “Custom”.
When set to “Fixed”, the phone will examine DHCP options 128,144, 157
and 191 (in that order) for a valid DVD string.
When set to "Custom", the value set in "VLAN ID Option" will be
examined for a valid DVD string.
DVD string in the DHCP option must meet the following conditions to be
valid:
— Must start with ?VLAN-A=? (case-sensitive)
— Must contain at least one valid ID
— VLAN IDs range from 0 to 4095
— Each VLAN ID must be separated by a ?+? character
— The string must be terminated by a ?;?
— All characters after the ?;? will be ignored
— There must be no white space before the ?;?
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— VLAN IDs may be decimal, hex, or octal
For example:
The following DVD strings will result in the phone using VLAN 10:
VLAN-A=10;
VLAN-A=0x0a;
VLAN-A=012;
If a VLAN tag is assigned by CDP, DHCP VLAN tags will be ignored.
Note
The following figure shows the phone’s processing to determine if the VLAN
ID is valid:
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Administrator’s Guide SoundPoint IP / SoundStation IP
Parsing Vendor ID Information
After the phone boot, it sends a DHCP Discover packet to the DHCP server.
This is found in the Bootstrap Protocol/option "Vendor Class Identifier"
section of the packet and includes the phone’s part number and the bootROM
version. The format of this option's data is not specified in RFC 2132, but is left
to each vendor to define its own format. To be useful, every vendor's format
must be distinguishable from every other vendor's format. To make our
format uniquely identifiable, the format follows RFC 3925, which uses the
IANA Private Enterprise number to determine which vendor's format should
be used to decode the remaining data. The private enterprise number assigned
to Polycom is 13885 (0x0000363D).
This vendor ID information is not a character string, but an array of binary
data. The steps for parsing are as follows:
1. Check for the Polycom signature at the start of the option:
4 octet: 00 00 36 3d
2. Get the length of the entire list of sub-options:
1 octet
3. Read the field code and length of the first sub-option, 1+1 octets
4. If this is a field you want to parse, save the data.
5. Skip to the start of the next sub-option.
6. Repeat steps 3 to 5 until you have all the data or you encounter the
End-of-Suboptions code (0xFF).
For example, the following is a sample decode of a packet from an IP601:
3c 74
- Option 60, length of Option data (part of the DHCP spec.)
00 00 36 3d
- Polycom signature (always 4 octects)
6f
- Length of Polycom data
01 07 50 6f 6c 79 63 6f 6d
- sub-option 1 (company), length, "Polycom"
02 15 53 6f 75 6e 64 50 6f 69 6e 74 49 50 2d 53 50 49 50 5f 36 30 31
- sub-option 2 (part), length, "SoundPointIP-SPIP_601"
03 10 32 33 34 35 2d 31 31 36 30 35 2d 30 30 31 2c 32
- sub-option 3 (part number), length, "2345-11605-001,2"
04 1c 53 49 50 2f 54 69 70 2e 58 58 58 58 2f 30 38 2d 4a 75 6e 2d 30 37
20 31 30 3a 34 34
- sub-option 4 (Application version), length, "SIP/Tip.XXXX/08-Jun-07
10:44"
05 1d 42 52 2f 33 2e 31 2e 30 2e 58 58 58 58 2f 32 38 2d 41 70 72 2d 30
35 20 31 33 3a 33 30
- sub-option 5 (BootROM version), length, "BR/3.1.0.XXXX/28-Apr-05
13:30"
ff
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- end of sub-options
For the BootROM, sub-option 4 and sub-option 5 will contain the same string.
The string is formatted as follows:
<apptype>/<buildid>/<date+time>
where:
<apptype> can be 'BR' (BootROM) or 'SIP' (SIP Application)
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D
Third Party Software
This appendix provides the copyright statements for third party software
products that are part of the Session Initiation Protocol (SIP) application.
Ares
Copyright 1998 by the Massachusetts Institute of Technology.
Permission to use, copy, modify, and distribute this software and its
documentation for any purpose and without fee is hereby granted, provided
that the above copyright notice appear in all copies and that both that
copyright notice and this permission notice appear in supporting
documentation, and that the name of M.I.T. not be used in advertising or
publicity pertaining to distribution of the software without specific, written
prior permission.
M.I.T. makes no representations about the suitability of this software for any
purpose. It is provided "as is" without express or implied warranty.
OpenSSL
The OpenSSL toolkit stays under a dual license, i.e. both the conditions of the
OpenSSL License and the original SSLeay license apply to the toolkit. See
below for the actual license texts. Actually both licenses are BSD-style Open
Source licenses. In case of any license issues related to OpenSSL please contact
OpenSSL License
Copyright (c) 1998-2003 The OpenSSL Project. All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are met:
1. Redistributions of source code must retain the above copyright notice, this
list of conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
and/or other materials provided with the distribution.
3. All advertising materials mentioning features or use of this software must
display the following acknowledgment:
"This product includes software developed by the OpenSSL Project for use in
the OpenSSL Toolkit. (http://www.openssl.org/)"
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4. The names "OpenSSL Toolkit" and "OpenSSL Project" must not be used to
endorse or promote products derived from this software without prior written
permission. For written permission, please contact [email protected].
5. Products derived from this software may not be called "OpenSSL" nor may
"OpenSSL" appear in their names without prior written permission of the
OpenSSL Project.
6. Redistributions of any form whatsoever must retain the following
acknowledgment:
"This product includes software developed by the OpenSSL Project for use in
the OpenSSL Toolkit (http://www.openssl.org/)"
THIS SOFTWARE IS PROVIDED BY THE OpenSSL PROJECT ``AS IS'' AND
ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND
FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
EVENT SHALL THE OpenSSL PROJECT OR ITS CONTRIBUTORS BE
LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER
IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
This product includes cryptographic software written by Eric Young
Original SSLeay License:
Copyright (C) 1995-1998 Eric Young ([email protected])
All rights reserved.
This package is an SSL implementation written by Eric Young
The implementation was written so as to conform with Netscape’s SSL.
This library is free for commercial and non-commercial use as long as the
following conditions are adhered to. The following conditions apply to all
code found in this distribution, be it the RC4, RSA, lhash, DES, etc., code; not
just the SSL code. The SSL documentation included with this distribution is
covered by the same copyright terms except that the holder is Tim Hudson
Copyright remains Eric Young's, and as such any Copyright notices in the
code are not to be removed. If this package is used in a product, Eric Young
should be given attribution as the author of the parts of the library used. This
can be in the form of a textual message at program startup or in documentation
(online or textual) provided with the package. Redistribution and use in
source and binary forms, with or without modification, are permitted
provided that the following conditions are met:
1. Redistributions of source code must retain the copyright notice, this list of
conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright notice,
this list of conditions and the following disclaimer in the documentation
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Third Party Software
and/or other materials provided with the distribution.
3. All advertising materials mentioning features or use of this software must
display the following acknowledgement: "This product includes
cryptographic software written by Eric Young ([email protected])"
The word 'cryptographic' can be left out if the routines from the library being
used are not cryptographic related.
4. If you include any Windows specific code (or a derivative thereof) from the
apps directory (application code) you must include an acknowledgement:
"This product includes software written by Tim Hudson ([email protected])"
THIS SOFTWARE IS PROVIDED BY ERIC YOUNG ``AS IS'' AND ANY
EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL
THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT,
INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY
WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
The licence and distribution terms for any publicly available version or
derivative of this code cannot be changed. i.e. this code cannot simply be
copied and put under another distribution licence [including the GNU Public
Licence.]
zlib
(C) 1995-2002 Jean-loup Gailly and Mark Adler
This software is provided 'as-is', without any express or implied warranty. In
no event will the authors be held liable for any damages arising from the use
of this software. Permission is granted to anyone to use this software for any
purpose, including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not claim
that you wrote the original software. If you use this software in a product, an
acknowledgment in the product documentation would be appreciated but is
not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
Jean-loup Gailly
Mark Adler
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Expat:
Copyright (c) 1998, 1999, 2000 Thai Open Source Software Center Ltd and
Clark Cooper
Permission is hereby granted, free of charge, to any person obtaining a copy of
this software and associated documentation files (the "Software"), to deal in
the Software without restriction, including without limitation the rights to use,
copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the
Software, and to permit persons to whom the Software is furnished to do so,
subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY
KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE
WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR
PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM,
DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
DEALINGS IN THE SOFTWARE.
curl
COPYRIGHT AND PERMISSION NOTICE
Copyright (c) 1996 - 2004, Daniel Stenberg, <[email protected]>.
All rights reserved.
Permission to use, copy, modify, and distribute this software for any purpose
with or without fee is hereby granted, provided that the above copyright
notice and this permission notice appear in all copies.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY
KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE
WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR
PURPOSE AND NONINFRINGEMENT OF THIRD PARTY RIGHTS. IN NO
EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE
FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN
ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT
OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
DEALINGS IN THE SOFTWARE.
Except as contained in this notice, the name of a copyright holder shall not be
used in advertising or otherwise to promote the sale, use or other dealings in
this Software without prior written authorization of the copyright holder.
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Third Party Software
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Administrator’s Guide SoundPoint IP / SoundStation IP
D - 6
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Index
A
IP TOS call control A–50
keys A–63
acoustic echo cancellation 4–44
acoustic echo cancellation <aec> A–40
acoustic echo suppression <aes> A–41
alert information A–14
local protocol A–6
localization A–19
main browser A–81
Microbrowser A–79
multilingual A–20
network monitoring A–51
outbound proxy A–13
password lengths A–74
platform A–65
animations <anim> A–66
application configuration
acoustic echo cancellation A–40
acoustic echo suppression A–41
animations A–66
port A–54
presence A–60
protocol A–6
protocol server A–7
protocol special events A–15
provisioning A–75
RAM disk A–76
automatic gain control A–43
background noise suppression A–42
bitmaps A–65
bulk drive A–82
call handling configuration A–55
call progress patterns A–30
chord-sets A–26
receive equalization A–43
request A–76
request delay A–77
request validation A–14
resource A–78
codec preferences A–35
codec profiles A–36
conference setup A–15
date and time A–23
ring type A–33
routing server A–19
sampled audio for sound effects A–27
SDP A–9
dial plan A–16
dial plan, emergency A–19
directory A–58
dual tone multi-frequency A–25
encryption A–74
security A–73
shared calls A–57
Ethernet call control A–48
event logging A–69
SIP A–10
sound effect patterns A–29
sound effects A–28
tones A–24
feature A–77
finder A–78
fonts A–60
transmit equalization A–45
USB port A–83
gains A–37
graphic icons A–68
user preferences A–23
voice activity detection A–47
voice coding algorithms
voice coding algorithms <codecs> A–34
voice settings A–34
volume persistence A–37
web server A–54
hold, local reminder A–58
idle display A–81
indicator classes A–67
indicator patterns A–67
indicators, assignments A–67
Index – 1
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Administrator’s Guide SoundPoint IP / SoundStation IP
application configuration file A–4
application error messages 5–3
busy <busy> A–91
busy lamp field 4–25
application files
overview 2–6
C
Applications key 4–28
attendant <attendant> A–97
attended transfers 4–17
call control <callControl> A–48
call control, third party B–9
call handling configuration <call> A–55
call hold 4–16
call log 4–3
call park/retrieve 4–20
call progress patterns A–30
call progress tones, synthesized 4–28
call timer 4–3
audio codecs 4–45
automatic call distribution 4–33
automatic gain control 4–46
automatic gain control <agc> A–43
automatic off-hook call placement 4–16
automatic off-hook call placement
<autoOffHook> A–88
call transfer 4–17
call waiting 4–3
B
background logo
called party identification 4–4
calling party identification 4–4
calls <calls> A–88
central provisioning, overview 2–6
changing the key on the phone C–5
chord-sets <chord> A–26
codec preferences <codecPref> A–35
codec profiles <audioProfile> A–36
comfort noise fill 4–46
conference setup <conference> A–15
configurable feature keys 4–21
configuration file encryption 4–49
configuration file example 4–39
adding C–5
configuration file changes C–6
background noise suppression 4–46
background noise suppression <ns> A–42
basic logging A–71
basic protocols
header support B–4
hold implementation B–9
request support B–3
response support B–6
RFC and Internet draft support B–2
transfer B–9
basic TCP/IP A–50
blind transfers 4–17
boot failure messages 5–7
boot server security policy 3–14
configuring SoundPoint IP / SoundStation IP
phones locally 4–50
connected party identification 4–5
consultative transfers 4–17
context sensitive volume control 4–5
custom certificates 4–48
customizable audio sound effects 4–5
customizable fonts and indicators 4–26
boot servers
deploying phones 3–15
redundant 3–12
security policy 3–14
setting up 3–13
bootROM 2–3
D
bootROM and application wrapper 2–5
bootROM error messages 5–2
bootROM tasks 2–3
date and time <datetime> A–23
default feature key layouts C–10
deploying phones from the boot server 3–14
device <device> A–100
DHCP
secondary server 3–3
DHCP INFORM 3–3
bootROM/SIP application dependencies C–7
bridged line appearance signaling B–10
bridged line appearances 4–24
browser limits A–81
bulk drive <bulkDrive> A–82
Index – 2
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Index
DHCP menu 3–7
H
DHCP or manual TCP/IP setup 3–2
dial plan <dialplan> A–16
handset, headset, and speakerphone 4–9
hold <hold> A–58
digit map
default A–18
I
examples A–17
match and replace A–17
idle display <idleDisplay> A–80
idle display animation 4–14
incoming signaling validation 4–49
indicator classes <class> A–67
indicators A–65
assignments A–67
installing SIP application 3–14
instant messaging 4–26
IP TOS A–48
IP TOS call control <callControl> A–50
IP_400 font A–62
digit map <digitmap> A–94
directed call pick-up 4–19
directory <dir> A–58
distinctive call waiting 4–7
distinctive incoming call treatment 4–6
distinctive ringing 4–7
diversion A–90
DNS SIP server name resolution 4–35
do not disturb 4–8
downloadable fonts 4–28
IP_500 font A–63
IP_600 font A–63
DTMF event RTP payload 4–44
DTMF tone generation 4–43
dual tone multi-frequency <DMTF> A–25
J
jitter buffer 4–42
K
key features 1–5
keys <key> A–63
E
encryption <encryption> A–74
Ethernet IEEE 802.1p/Q A–47
Ethernet menu 3–11
L
language support 4–27
languages, adding new A–21
last call return 4–20
LEDs A–68
F
length <length> A–74
local / centralized conferencing 4–17
local <local> A–6
local contact directory 4–9
local contact directory file format 4–10
local digit map 4–12
local reminder <localReminder> A–58
local user and administrator privilege levels 4–48
localization <lcl> A–19
feature <feature> A–77
features
list of 1–5
finder <finder> A–78
flash parameter configuration A–99
fonts <font> A–60
forward all <fwd> A–90
log files 5–4
logging <log> A–69
low-delay audio packet transmission 4–42
G
gains <gain> A–37
graphic icons <gi> A–68
group call pick-up 4–20
Index – 3
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Administrator’s Guide SoundPoint IP / SoundStation IP
M
patterns <pat> A–29
patterns <pattern> A–67
per-phone configuration
attendant A–97
MAC address
definition A–2
automatic off-hook call placement A–88
busy A–91
main browser <main> A–82
main menu 3–6
manual configuration, overview 2–7
manual log upload 5–6
calls A–87
dial plan, emergency A–95
digit map A–92
forward all A–90
message waiting indicator A–96
messaging A–95
missed call configuration A–88
Network Address Translation A–96
no answer A–91
quotas A–79
registration A–83
master configuration file
model number version A–4
part number substitution A–4
master configuration files
details A–2
overview 2–5
message waiting indication 4–6
message waiting indicator <mwi> A–96
messaging <msg> A–95
Microbrowser 4–28
Microbrowser <mb> A–79
microphone mute 4–13
roaming buddies A–98
roaming privacy A–98
routing A–94
routing server A–94
per-phone configuration file A–82
phone1.cfg A–82
port <port> A–54
Microsoft Live Communications Server 2005
Integration 4–38
migration dependencies C–8
miscellaneous patterns A–32
presence 4–37
missed call configuration <serverMissedCall>
presence <pres> A–60
protocol <voIpProt> A–6
protocol server <server> A–7
protocol special events <specialEvent> A–15
provisioning <prov> A–75
provisioning protocols 3–4
provisioning protocols, supported 3–4
missed call notification 4–4
model number substitution A–4
modifying network configuration 3–5
multilingual <ml> A–20
multilingual user interface 4–27
multiple call appearances 4–23
multiple line keys per registration 4–22
multiple registrations 4–31
Q
quotas <quotas> A–79
N
Network Address Translation <nat> A–96
network configuration, modifying 3–5
network monitoring <netMon> A–51
no answer <noanswer> A–91
R
RAM disk <ramdisk> A–76
receive equalization <rxEq> A–43
registration <reg> A–83
reliability of provisional responses B–9
request <request> A–76
request delay <delay> A–77
request validation <requestValidation> A–14
resetting to factory defaults 3–5
resource <res> A–78
O
Option 66 3–7
outbound proxy <outboundProxy> A–13
P
packet error concealment 4–42
password <pwd> A–74
Index – 4
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Index
resource files, overview 2–7
RFC support B–2
ring type <rt> A–33
ringer patterns A–31
roaming buddies <roaming_buddies> A–98
roaming privacy <roaming_provacy> A–98
routing <routing> A–94
instant messaging and presence leveraging
extensions B–9
RFC 2–1
SIP application
description 2–4
installing 3–14
upgrading 3–17
SIP basic protocols, header support B–4
sip.cfg A–4
SIP<SIP> A–10
sound effects <se> A–28
S
SoundPoint IP / SoundStation IP phones
features, overview 2–8
introduction 1–1
network 2–2
SoundPoint IP 330
switching text entry mode 3–7
SoundPoint IP desktop phones 1–2
features, list of 1–5
SoundStation IP conference phone 1–4
SoundStation IP conference phones
features, list of 1–5
speed dial 4–13
status menu 5–4
sampled audio files A–28
sampled audio for sound effects <saf> A–27
scheduled logging parameters A–72
SDP <SDP> A–9
security <sec> A–73
server menu 3–9
server redundancy 4–34
forwarding
T
setting up
text entry mode, switching 3–7
time and date display 4–13
time synchronization A–51
transmit equalization <txEq> A–45
advanced features 4–20
audio features 4–42
basic features 4–1
boot server 3–12
network 3–2
troubleshooting
security features 4–47
application error messages 5–3
application logging options 5–5
audio issues 5–14
shared call appearance signaling B–10
shared calls <shared> A–57
blinking time 5–4
SIP
boot failure messages 5–7
bootROM error messages 5–2
calling issues 5–12
1xx Responses - Provisional B–6
2xx Responses - Success B–6
3xx Responses - Redirection B–7
4xx Responses - Request Failure B–7
5xx Responses - Server Failure B–8
6xx Responses - Global Failure B–8
application architecture 2–3
basic protocols, hold implementation B–9
basic protocols, request support B–3
basic protocols, response support B–6
basic protocols, RFC and Internet draft
support B–2
Config file error. Error is 5–3
controls issues 5–10
Could not contact boot server 5–2
displays issues 5–13
Error loading 5–3
Error, application is not present! 5–2
Failed to get boot parameters via DHCP 5–2
log files 5–4
manual log upload 5–6
basic protocols, transfer B–9
Index – 5
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Administrator’s Guide SoundPoint IP / SoundStation IP
Network link is down 5–3
Not all configuration files were present 5–3
power and startup issues 5–9
reading a boot log 5–7
reading an application log 5–8
registration status 5–3
scheduled logging 5–6
screens and systems access issues 5–11
trusted certificate authority list C–1
U
upgrading SIP application 3–17
USB port <usb> A–82
user interface, soft key activated 4–13
user preferences <up> A–23
V
VLAN ID using DHCP C–14
voice activity detection 4–43
voice activity detection <vad> A–47
voice mail integration 4–30
voice setting <voice> A–34
volume persistence <volume> A–37
W
web server <httpd> A–54
Index – 6
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Addendum to SIP 2.2 Administrator’s
Guide
This addendum addresses changes to the SoundPoint IP / SoundStation IP
SIP 2.2 Administrator’s Guide specific to the release of the SoundPoint IP 560
desktop phone.
The SoundPoint IP 560 desktop phone hardware behaves in a similar manner
to the SoundPoint IP 550 except for:
•
The SoundPoint IP 560 features a future-proof dual-port Gigabit Ethernet
switch for seamless integration with a computer or desktop server.
The new or changed features include:
•
•
•
•
For more information, refer to the Release Notes for the SIP Application,
Version 2.2.2 .
For more information on the SoundPoint IP 560 desktop phone, refer to the User
Note
New or Changed Features
Ethernet Menu
The SoundPoint IP 560 phone has an additional LAN port mode of 1000FD and
an additional PC port mode of 1000FD.
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Administrator’s Guide for the SoundPoint IP 560
Distribution Zip File
There is a separate sip.ld file in the archive for the SoundPoint IP 560 phone
called 2345-12560-001.sip.ld .
Local Contact Directory
The local contact directory can be configured to be read only.
A new configuration attribute can be found in the <dir/> parameter in the
sip.cfg configuration file. It is defined as follows:
Permitted
Attribute
Values
Default
Interpretation
dir.local.readonly
0, 1
1
Specifies whether or not local
contact directory is read only.
If set to 0 or Null, the local contact
directory is editable.
If set to 1, the local contact directory
is read only.
LCD Backlight
Backlight intensity configuration on the SoundPoint IP 560 phone has an
additional menu option. Users can now modify the maximum backlight
intensity.
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