Polycom SOUNDPOINT SIP 2.2.0 User Manual

Administrator’s Guide for the

SoundPoint

®

IP/SoundStation

®

IP

Family

SIP 2.2.0

August, 2007 Edition

1725-11530-220 Rev. A

SIP 2.2.0

Trademark Information

Polycom®, the Polycom logo design, SoundPoint® IP, SoundStation®, SoundStation VTX 1000®, ViaVideo®,
ViewStation®, and Vortex® are registered trademarks of Polycom, Inc. Conference Composer™, Global Management
System™, ImageShare™, Instructor RP™, iPower™, MGC™, PathNavigator™, People+Content™, PowerCam™,

2

Pro-Motion™, QSX™, ReadiManager™, Siren™, StereoSurround™, V

IU™, Visual Concert™, VS4000™, VSX™, and
the industrial design of SoundStation are trademarks of Polycom, Inc. in the United States and various other countries.
All other trademarks are the property of their respective owners.

Patent Information

The accompanying product is protected by one or more U.S. and foreign patents and/or pending patent applications
held by Polycom, Inc.

Disclaimer

Some countries, states, or provinces do not allow the exclusion or limitation of implied warranties or the limitation of
incidental or consequential damages for certain products supplied to consumers, or the limitation of liability for personal
injury, so the above limitations and exclusions may be limited in their application to you. When the implied warranties
are not allowed to be excluded in their entirety, they will be limited to the duration of the applicable written warranty. This
warranty gives you specific legal rights which may vary depending on local law.

Copyright Notice

Portions of the software contained in this product are:
Copyright © 1998, 1999, 2000 Thai Open Source Software Center Ltd. and Clark Cooper
Copyright © 1998 by the Massachusetts Institute of Technology
Copyright © 1998-2003 The OpenSSL Project
Copyright © 1995-1998 Eric Young ([e-mail]). All rights reserved
Copyright © 1995-2002 Jean-Loup Gailly and Mark Adler
Copyright © 1996 - 2004, Daniel Stenberg, <[e-mail]>

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 include d 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.

© 2007 Polycom, Inc. All rights reserved.
Polycom Inc.

4750 Willow Road
Pleasanton, CA 94588-2708
USA

No part of this document may be reproduced or transmitted in any form or by any means, electronic or mechanical, for
any purpose, without the express written permission of Polycom, Inc. Under the law, reproducing includes translating
into another language or format.

As between the parties, Polycom, Inc. retains title to, and ownership of, all proprietary rights with respect to the software
contained within its products. The software is protected by United States copyright laws and international treaty
provision. Therefore, you must treat the software like any other copyrighted material (e.g. a book or sound recording).

Every effort has been made to ensure that the information in this manual is accurate. Polycom, Inc. is not responsible
for printing or clerical errors. Information in this document is subject to change without notice.

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
Technical Support at http://www.polycom.com/support/voice/.

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: _________________________________________________

®

reselle r or go to Polycom

iii

Administrator’s Guide SoundPoint IP / SoundStation IP

iv

Contents

About This Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . iii

1 Introducing the SoundPoint IP / SoundStation IP Family . . . 1–1

2 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–1

SoundPoint IP Desktop Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1–2

SoundStation IP Conference Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1–4

Key Features of Your SoundPoint IP / SoundStation IP Phones . . . . . . . 1–5

Where SoundPoint IP / SoundStation IP Phones Fit . . . . . . . . . . . . . . . . . 2–2

Session Initiation Protocol Application Architecture . . . . . . . . . . . . . . . . . 2–3

BootROM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–3

Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–4

Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–5

Resource Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–7

Available Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–8

3 Setting up Your System . . . . . . . . . . . . . . . . . . . . . . . . . . .3–1

Setting Up the Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–2

DHCP or Manual TCP/IP Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–2

Supported Provisioning Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–4

Modifying the Network Configuration . . . . . . . . . . . . . . . . . . . . . . . . . 3–5

Setting Up the Boot Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–12

Deploying Phones From the Boot Server . . . . . . . . . . . . . . . . . . . . . . . . . . 3–14

Upgrading SIP Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–17

Supporting SoundPoint IP and SoundStation IP Phones . . . . . . . . . 3–17

Supporting SoundPoint IP 300 and 500 Phones . . . . . . . . . . . . . . . . . 3–18

4 Configuring Your System . . . . . . . . . . . . . . . . . . . . . . . . . . 4–1

Setting Up Basic Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–1

Call Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–3

Call Timer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–3

v

Administrator’s Guide SoundPoint IP / SoundStation IP

Call Waiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–3

Called Party Identification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–4

Calling Party Identification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–4

Missed Call Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–4

Connected Party Identification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–5

Context Sensitive Volume Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–5

Customizable Audio Sound Effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–5

Message Waiting Indication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–6

Distinctive Incoming Call Treatment . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–6

Distinctive Ringing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–7

Distinctive Call Waiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–7

Do Not Disturb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–8

Handset, Headset, and Speakerphone . . . . . . . . . . . . . . . . . . . . . . . . . 4–9

Local Contact Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–9

Local Digit Map . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–12

Microphone Mute . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–13

Soft Key Activated User Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–13

Speed Dial . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–13

Time and Date Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–13

Idle Display Animation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–14

Ethernet Switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–15

Automatic Off-Hook Call Placement . . . . . . . . . . . . . . . . . . . . . . . . . . 4–16

Call Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–16

Call Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–17

Local / Centralized Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . 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

Configurable Feature Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–21

Multiple Line Keys per Registration . . . . . . . . . . . . . . . . . . . . . . . . . . 4–22

Multiple Call Appearances . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–23

Shared Call Appearances . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–23

Bridged Line Appearance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–24

Busy Lamp Field . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–25

Customizable Fonts and Indicators . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–26

Instant Messaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–26

Multilingual User Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–27

vi

Contents

Downloadable Fonts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–28

Synthesized Call Progress Tones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–28

Microbrowser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–28

Real-Time Transport Protocol Ports . . . . . . . . . . . . . . . . . . . . . . . . . . 4–29

Network Address Translation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–30

Voice Mail Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–30

Multiple Registrations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–31

Automatic Call Distribution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–33

Server Redundancy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–34

Presence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–37

Microsoft Live Communications Server 2005 Integration . . . . . . . . 4–38

Setting Up Audio Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–42

Low-Delay Audio Packet Transmission . . . . . . . . . . . . . . . . . . . . . . . 4–42

Jitter Buffer and Packet Error Concealment . . . . . . . . . . . . . . . . . . . . 4–42

Voice Activity Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–43

DTMF Tone Generation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–43

DTMF Event RTP Payload . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–44

Acoustic Echo Cancellation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–44

Audio Codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–45

Background Noise Suppression . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–46

Comfort Noise Fill . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–46

Automatic Gain Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–46

IP Type-of-Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–46

IEEE 802.1p/Q . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–47

Setting Up Security Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–47

Local User and Administrator Privilege Levels . . . . . . . . . . . . . . . . . 4–48

Custom Certificates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–48

Incoming Signaling Validation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–49

Configuration File Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–49

Configuring SoundPoint IP / SoundStation IP Phones Locally . . . . . . . 4–50

5 Troubleshooting Your SoundPoint IP / SoundStation IP

Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–1

Error Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–2

BootROM Error Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–2

Application Error Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–3

Status Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–4

Log Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–4

Reading a Boot Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–7

Reading an Application Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–8

vii

Administrator’s Guide SoundPoint IP / SoundStation IP

Power and Startup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–9

Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–10

Access to Screens and Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–11

Calling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–12

Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–13

Audio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–14

Upgrading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–14

A Configuration Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .A–1

Master Configuration Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–2

Application Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–4

Protocol <volpProt/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–6

Dial Plan <dialplan/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–16

Localization <lcl/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–19

User Preferences <up/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–23

Tones <tones/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–24

Sampled Audio for Sound Effects <saf/> . . . . . . . . . . . . . . . . . . . . . A–27

Sound Effects <se/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–28

Voice Settings <voice/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–34

Quality of Service <QOS/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–47

Basic TCP/IP <TCP_IP/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–50

Web Server <httpd/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–54

Call Handling Configuration <call/> . . . . . . . . . . . . . . . . . . . . . . . . A–55

Directory <dir/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–58

Presence <pres/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–60

Fonts <font/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–60

Keys <key/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–63

Bitmaps <bitmap/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–65

Indicators <ind/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–65

Event Logging <log/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–69

Security <sec/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–73

License <license/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–74

Provisioning <prov/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–75

RAM Disk <ramdisk/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–76

Request <request/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–76

Feature <feature/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–77

Resource <res/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–78

Microbrowser <mb/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–79

USB Port <usb/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–83

viii

Contents

Per-Phone Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–82

Registration <reg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–83

Calls <call/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–87

Diversion <divert/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–90

Dial Plan <dialplan/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–92

Messaging <msg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–95

Network Address Translation <nat/> . . . . . . . . . . . . . . . . . . . . . . . A–96

Attendant <attendant/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–97

Roaming Buddies <roaming_buddies/> . . . . . . . . . . . . . . . . . . . . . A–98

Roaming Privacy <roaming_privacy/> . . . . . . . . . . . . . . . . . . . . . . A–98

Flash Parameter Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–99

B Session Initiation Protocol (SIP) . . . . . . . . . . . . . . . . . . . . . B–1

RFC and Internet Draft Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–2

Request Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–3

Header Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–4

Response Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–6

Hold Implementation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–9

Reliability of Provisional Responses . . . . . . . . . . . . . . . . . . . . . . . . . . . B–9

Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–9

Third Party Call Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B–9

SIP for Instant Messaging and Presence Leveraging Extensions . . . B–9

Shared Call Appearance Signaling . . . . . . . . . . . . . . . . . . . . . . . . . . . B–10

Bridged Line Appearance Signaling . . . . . . . . . . . . . . . . . . . . . . . . . . B–10

C Miscellaneous Administrative Tasks . . . . . . . . . . . . . . . . . .C–1

Trusted Certificate Authority List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–1

Encrypting Configuration Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–3

Changing the Key on the Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–5

Adding a Background Logo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–5

BootROM/SIP Application Dependencies . . . . . . . . . . . . . . . . . . . . . . . . C–7

Migration Dependencies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–8

Multiple Key Combinations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–9

Default Feature Key Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–10

Assigning a VLAN ID Using DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–14

Parsing Vendor ID Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–16

ix

Administrator’s Guide SoundPoint IP / SoundStation IP

D Third Party Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . .D–1

Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .Index–1

x

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.

1

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:

• SoundPoint IP Desktop Phones

— IP 301

— IP 320/330

— IP 430

— IP 501

— IP 550

— IP 600/601

— IP 650

• SoundStation IP Conference Phone

— IP 4000

This chapter also lists the key features available on the SoundPoint IP /
SoundStation IP phones running the latest software.

1 - 1

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

1 - 2

• SoundPoint IP 430

• SoundPoint IP 501

Introducing the SoundPoint IP / SoundStation IP Family

• SoundPoint IP 550

1 - 3

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.

1 - 4

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

1 - 5

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

1 - 6

Overview

2

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.

For more information on IP PBX and softswitch vendors, go to

http://www.polycom.com/techpartners1/ .

This chapter contains information on:

• Where SoundPoint IP / SoundStation IP Phones Fit

• Session Initiation Protocol Application Architecture

• Available Features

To install your SoundPoint IP / SoundStation IP phones on the network, refer
to Setting up Your System on page 3-1. To configure your SoundPoint IP /
SoundStation IP phones with the desired features, refer to Configuring Your

2 - 1

Administrator’s Guide SoundPoint IP / SoundStation IP

System on page 4-1. To troubleshoot any problems with your SoundPoint IP /

SoundStation IP phones on the network, refer to Troubleshooting Your

SoundPoint IP / SoundStation IP Phones on page 5-1.

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.

2 - 2

Session Initiation Protocol Application Architecture

Configuration

Resource

Files

bootROM

Application

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

Overview

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.

2 - 3

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.

Application

Note

Warning

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.

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.

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.
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.

2 - 4

Note

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.

Configuration

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.

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.

Warning

The phone configuration files consist of:

• Master Configuration Files

• Application Configuration Files

Configuration files should only be modified by a knowledgeable system
administrator. Applying incorrect parameters may render the phone unusable. The
configuration files which accompany a specific release of the SIP software must be
used together with that software. Failure to do this may render the phone unusable.

Master Configuration Files

The master configuration files can be one of:

• Specified master configuration file

• Per-phone master configuration file

• Default master configuration file

For more information, refer to Master Configuration Files on page A-2.

2 - 5

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.

Note

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.

2 - 6

Overview

The following figure shows one possible layout of the central provisioning
method.

Resource Files

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.

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

2 - 7

Administrator’s Guide SoundPoint IP / SoundStation IP

• Ring tones

• Synthesized tones

• Contact directories

Available Features

Note

Any new features introduced after SIP 2.1.2 are not supported on the
SoundPoint IP 300 and 500.

This section provides information the features available on the SoundPoint IP
/ SoundStation IP phones:

• Basic Features

— Automatic Off-Hook Call Placement—Supports an optional

automatic off-hook call placement feature for each .

— Call Forward—Provides a flexible call forwarding feature to forward

calls to another destination.

— Call Hold—Pauses activity on one call so that the user may use the

phone for another task, such as making or receiving another call.

— Call Log—Contains call information such as remote party

identification, time and date, and call duration in three separate lists,
missed calls, received calls, and placed calls on most platforms.

— Call Park/Retrieve—An active call can be parked. A parked call can

be retrieved by any phone.

— Call Timer—A separate call timer, in hours, minutes, and seconds, is

maintained for each distinct call in progress.

— Call Transfer—Call transfer allows the user to transfer a call in

progress to some other destination.

2 - 8

— 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 display and a configurable sound effect will be mixed with the
active call audio.

— Called Party Identification—The phone displays and logs the identity

of the party specified for outgoing calls.

— Calling Party Identification—The phone displays the caller identity,

derived from the network signalling, when an incoming call is
presented, if information is provided by the call server.

— Connected Party Identification—The identity of the party to which the

user has connected is displayed and logged, if the name is provided
by the call server.

Overview

— Context Sensitive Volume Control—The volume of user interface

sound effects, such as the ringer, and the receive volume of call audio
is adjustable.

— Customizable Audio Sound Effects—Audio sound effects used for

incoming call alerting and other indications are customizable.

— Directed Call Pick-Up and Group Call Pick-Up—Calls to another

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.

— Distinctive Call Waiting—Calls can be mapped to distinct call waiting

types.

— Distinctive Incoming Call Treatment—The phone can automatically

apply distinctive treatment to calls containing specific attributes.

— Distinctive Ringing—The user can select the ring type for each line

and the ring type for specific callers can be assigned in the contact
directory.

— Do Not Disturb—A do-not-disturb feature is available to temporarily

stop all incoming call alerting.

— Handset, Headset, and Speakerphone—SoundPoint IP phones come

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.

— 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.

— Last Call Return—The phone allows call server-based last call return.

— Local / Centralized Conferencing—The phone can conference

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.

— Local Contact Directory—The phone maintains a local contact

directory that can be downloaded from the boot server and edited
locally.

— Local Digit Map—The phone has a local digit map to automate the

setup phase of number-only calls.

— Message Waiting Indication—The phone will flash a message-waiting

indicator (MWI) LED when instant messages and voice messages are
waiting.

— Microphone Mute—When the microphone mute feature is activated,

visual feedback is provided.

2 - 9

Administrator’s Guide SoundPoint IP / SoundStation IP

— Missed Call Notification—The phone can display the number of calls

missed since the user last looked at the Missed Calls list.

— Soft Key Activated User Interface—The user interface makes

extensive use of intuitive, context-sensitive soft key menus.

— Speed Dial—The speed dial system allows calls to be placed quickly

from dedicated keys as well as from a speed dial menu.

— Time and Date Display—Time and date can be displayed in certain

operating modes such as when the phone is idle and during a call.

• Advanced Features

— Automatic Call Distribution—Supports ACD agent available and

unavailable and allows ACD login and logout. Requires call server
support.

— Bridged Line Appearance—Calls and lines on multiple phones can be

logically related to each other. Requires call server support.

— Busy Lamp Field—Allows monitoring the hook status and remote

party information of users through the busy lamp field (BLF) LEDs
and displays on an attendant console phone. Requires call server
support.

— Configurable Feature Keys—Certain key functions can be changed

from the factory defaults.

— 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.

— Downloadable Fonts—New fonts can be loaded onto the phone.

— Instant Messaging—Supports sending and receiving instant text

messages.

— Microbrowser—The SoundPoint IP 430, 501, 550, 600, 601, and 650

phones and the SoundStation IP 4000 phone support an XHTML
microbrowser.

— Microsoft Live Communications Server 2005

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.

— Multilingual User Interface—All phones except SoundPoint IP 301

have multilingual user interfaces.

— 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.

2 - 10

— Multiple Line Keys per Registration—More than one line key can be

allocated to a single .

Overview

— Multiple Registrations—SoundPoint IP phones support multiple s per

phone. (SoundStation IP 4000 supports a single .)

— Network Address Translation—The phones can work with certain

types of network address translation (NAT).

— Presence—Allows the phone to monitor the status of other

users/devices and allows other users to monitor it. Requires call
server support.

— Real-Time Transport Protocol Ports—The phone treats all real- time

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.

— 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.

— Shared Call Appearances—Calls and lines on multiple phones can be

logically related to each other. Requires call server support.

— 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. Customizable for certain regions, for
example, Europe has different tones from North America.

— Voice Mail Integration—Compatible with voice mail servers.

• Audio Features

— Acoustic Echo Cancellation—Employs advanced acoustic echo

cancellation for hands-free operation.

— Audio Codecs—Supports the standard audio codecs.

— Automatic Gain Control—Designed for hands-free operation, boosts

the transmit gain of the local user in certain circumstances.

— Background Noise Suppression—Designed primarily for hands-free

operation, reduces background noise to enhance communication in
noisy environments.

— Comfort Noise Fill—Designed to help provide a consistent noise level

to the remote user of a hands-free call.

— DTMF Event RTP Payload—Conforms to RFC 2833, which describes

a standard RTP-compatible technique for conveying DTMF dialing
and other telephony events over an RTP media stream.

— DTMF Tone Generation—Generates dual tone multi-frequency

(DTMF) tones in response to user dialing on the dial pad.

— IEEE 802.1p/Q—The phone will tag all Ethernet packets it transmits

with an 802.1Q VLAN header.

2 - 11

Administrator’s Guide SoundPoint IP / SoundStation IP

— IP Type-of-Service—Allows for the setting of TOS settings.

— Jitter Buffer and Packet Error Concealment—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.

— Low-Delay Audio Packet Transmission—Designed to minimize

latency for audio packet transmission.

— Voice Activity Detection—Conserves 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.

• Security Features

— Local User and Administrator Privilege Levels—Several local settings

menus are protected with two privilege levels, user and
administrator, each with its own password.

— Configuration File Encryption—Confidential information stored in

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.

— Custom Certificates—When trying to establish a connection to a boot

server for application provisioning, the phone trusts certificates
issued by widely recognized certificate authorities (CAs).

— Incoming Signaling Validation—Levels of security are provided for

validating incoming network signaling.

For more information on each feature and its associated configuration
parameters, see the appropriate section in Configuring Your System on page

4-1.

2 - 12

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:

• Setting Up the Network

• Setting Up the Boot Server

• Deploying Phones From the Boot Server

• Upgrading SIP Application

3

Note

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

System on page 4-1. For more information on the configuration files required

for setting up your system, refer to Configuration Files on page A-1.

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.

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.

3 - 1

Administrator’s Guide SoundPoint IP / SoundStation IP

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
to DHCP or Manual TCP/IP Setup on page 3-2.

The bootROM on the phone performs the provisioning functions of
downloading the bootROM, the <Ethernet address>.cfg file, and the SIP
application, and uploading log files. For more information, refer to Supported

Provisioning Protocols on page 3-4.

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

Modifying the Network Configuration on page 3-5.

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

Parameter

IP address 1•-- •
subnet mask 1•-- •

DHCP Option

DHCP

D priority when more than one source exists D
12 3 4

DHCP

Configuration File
(application only)

Local
FLASH

IP gateway 3•-- •

3 - 2

Setting up Your System

Alternate

Parameter

boot server
address

SIP server address
SNTP server

address
SNTP GMT offset 2•-• •
DNS server IP

address
alternate DNS

server IP address
DNS domain 15 • - - •

VLAN ID

DHCP Option

Refer to DHCP

Menu on page
3-7

151

Note: This value
is configurable.

42 then 4 • - • •

6•-- •

6•-- •

Refer to DHCP

Menu on page
3-7

DHCP

•• - •

•- - •

Warning: Cisco Discovery Protocol (CDP) overrides Local FLASH
that overrides DHCP VLAN Discovery.

DHCP

Configuration File
(application only)

Local
FLASH

Note

For more information on DHCP options, go to

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

tcpIpApp.sntp.address.overrideDHCP

A-51.

The CDP value can be obtained from a con nected Ethernet switch if the switch
supports CDP.

in Time Synchronization <sntp/> on page

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
discovering the provisioning server address is required. Connecting to a
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.

3 - 3

Administrator’s Guide SoundPoint IP / SoundStation IP

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

Server Menu on page 3-9). The boot server address can be an IP address,

domain string name, or URL. The boot server address can also be obtained
through DHCP. Configuration file names in the <Ethernet address>.cfg file
can include a transfer protocol, for example
https://usr:pwd@server/dir/file.cfg. If a user name and password are
specified as part of the server address or file name, they will be used only if the
server supports them.

Note

Note

A URL should contain forward slashes instead of back slashes and should not
contain spaces. Escape characters are not supported. If a user name and
password are not specified, the Server User and Server Password will be used
(refer to Server Menu on page 3-9).

Protocol used by
bootROM

301, 320, 330, 430,
Specified
Protocol

FTP FTP FTP
TFTP TFTP TFTP
HTTP HTTP HTTP
HTTPS 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.

501, 550, 600, 601,

650, 4000

Protocol used by
SIP Application

301, 320, 330, 430,
501, 550, 600, 601,
650, 4000

3 - 4

Setting up Your System

Note

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.

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

Certificate Authority List on page C-1).

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:

• Main Menu

• DHCP Menu

• Server Menu

• Ethernet Menu

• Syslog Menu

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

Multiple Key Combinations on page C-9.

3 - 5

Administrator’s Guide SoundPoint IP / SoundStation IP

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

discussed in DHCP or Manual TCP/IP Setup on page

3-2.

DHCP Menu Refer to DHCP Menu on page 3-7.

Note: Disabled when DHCP client is disabled.

Phone IP Address dotted-decimal IP address Phone’s IP address.

Note: Disabled when DHCP client is enabled.

Subnet Mask dotted-decimal subnet

mask

IP Gateway dotted-decimal IP address Phone’s default router.
Server Menu Refer to Server Menu on page 3-9.
SNTP Address dotted-decimal IP address

OR
domain name string

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

DNS Server dotted-decimal IP address Primary server to which the phone directs Domain

DNS Alternate Server dotted-decimal IP address Secondary server to which the phone directs Domain

DNS Domain domain name string Phone’s DNS domain.
Ethernet Refer to Ethernet Menu on page 3-11.
EM Power Enabled, Disabled This parameter is relevant if the phone gets Power over

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

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

Time (GMT) in half hour increments.

Name System (DNS) queries.

Name System queries.

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 Refer to Syslog Menu on page 3-11.

3 - 6

Setting up Your System

Note

Note

A parameter value of “???” indicates that the parameter has not yet been set and
saved in the phone’s configuration. Any such parameter should have its value set
before continuing.

The 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

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

Boot Server 0=Option 66 The phone will look for option number 66 (string type) in the

Values Description

messages before selecting an offer.

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
section, Server Menu. 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.

3 - 7

Administrator’s Guide SoundPoint IP / SoundStation IP

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
section, Server Menu.

3=Custom+Option 66The phone will first use the custom option if present or use

Boot Server Option 128 through 254

(Cannot be the
same as VLAN ID
Option)

Boot Server Option Type 0=IP Address,

1=String

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.

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.

When the Boot Server parameter is set to Custom, this
parameter specifies the type of the DHCP option in which the
phone will look for its boot server. The IP Address must specify
the boot server. The String must match one of the formats
described for Server Address in the following section, Server

Menu.

3 - 8

Name

Setting up Your System

Possible
Values Description

VLAN Discovery 0=Disabled

(default)
1=Fixed Use predefined DHCP vendor-specific option values of 128,

2=Custom Use the number specified in the VLAN ID Option field as th e

VLAN ID Option 128 through 254

(Cannot be the
same as Boot
Server Option)

(default is 129)

Note

If multiple alternate DHCP servers respond:

• The phone should gather the responses from alternate DHCP servers.

• If configured for

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.

No VLAN discovery through DHCP.

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

DHCP private option value.
The DHCP private option value (when VLAN Discovery is set

to Custom).
For more information, refer to Assigning a VLAN ID Using

DHCP on page C-14.

Custom+Option66

, the phone will select the first response that

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
server. Refer to Supported Provisioning Protocols on
page 3-4.

Note: Active FTP is not supported for bootROM version

3.0 or later. Passive FTP is still supported.

3 - 9

Administrator’s Guide SoundPoint IP / SoundStation IP

Name Possible Values Description

Server Address dotted-decimal IP address

OR
domain name string
OR
URL

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

The boot server to use if the DHCP client is disabled, the
DHCP server does not send a boot server option, or the
Boot Server parameter is set to Static. The phone can
contact multiple IP addresses per DNS name. These
redundant boot servers must all use the same protocol. If
a URL is used it can include a user name and password.
Refer to Supported Provisioning Protocols on page 3-4. A
directory and the master configuration file can be
specified.

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

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

(if required) for the selected Server Type.

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

Server Password any string The password used when the phone logs in to the server

if required for the selected Server Type.

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

File Transmit Tries 1 to 10

Default 3

The number of attempts to transfer a file. (An attempt is
defined as trying to download the file from all IP
addresses that map to a particular domain name.)

Retry Wait 0 to 300

Default 1

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

For more information, refer to Deploying Phones From the

Boot Server on page 3-14.

Provisioning
Method

Network Cable/DSL,

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

the normal process).

The network environment the phone is operating in.
LAN,
Dial-up

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.

3 - 10

Setting up Your System

Note

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.

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 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.

LAN Port Mode 0 = Auto

1 = 10HD
2 = 10FD
3 = 100HD
4 = 100FD

PC Port Mode Auto, 10HD, 10FD, 100HD,

100FD

Note

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.

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.

The network speed over the Ethernet.

The default value is Auto.

The network speed over the Ethernet.

The default value is Auto.

3 - 11

Administrator’s Guide SoundPoint IP / SoundStation IP

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 syslog protocol is defined in RFC 3164. For more information on syslog,
go to http://www.ietf.org/rfc/rfc3164.txt?number=3164 .

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

Server Type None=0,

UDP=1,
TCP=2,
TLS=3

Facility 0 to 23 A description of what generated the log message. For

Render Level 1 to 6 Specifies the lowest class of event that will be rendered to

Prepend MAC
Address

Enabled, Disabled If enabled, the phone’s MAC address is prepended to the

The syslog server IP address or host name.

The default value is NULL.

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.

more information, refer to section 4.1.1 of RFC 3164.

The default value is 16, which maps to “local 0”.

syslog. It is based on

lower value.

Refer to Basic Logging <level/><change/> and <render/>

on page A-71.

Note: Use left and right arrow keys to change values.

log message sent to the syslog server.

log.render.level

and can be a

Setting Up the Boot Server

3 - 12

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
protocol used on each platform, refer to Supported Provisioning Protocols on
page 3-4.

Setting up Your System

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 Menu on page 3-9 can be used to configure the number of times each

server will be tried for a file transfer and also how long to wait between each
attempt. The maximum number of servers to be tried is configurable. For more
information, contact your Certified Polycom Reseller.

Note

Note

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

If you want to use redundancy for uploads, 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.

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.

Note

Note

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.

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.

3 - 13

Administrator’s Guide SoundPoint IP / SoundStation IP

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
LOG_FILE_DIRECTORY, refer to Master Configuration Files on page A-2.)

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.

3 - 14

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:

Note

Note

Note

For more information on encrypting configuration files, refer to Encrypting

Configuration Files on page C-3.

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 Obtain a list of phone Ethernet addresses (barcoded label on

underside of phone and on the outside of the box).

b Create per-phone phone[MACaddress].cfg file by using the

phone1.cfg file from the distribution as templates.

For more information on the phone1.cfg file, refer to Per-Phone

Configuration on page A-82.

Throughout this guide, the terms Ethernet address and MAC address are used
interchangeable.

c Edit contents of phone[MACaddress].cfg if desired.

Note

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 Phone s” whitepaper at

www.polycom.com/support/voice/ .

For more information, especially on the SIP server address, refer to SIP

<SIP/> on page A-10.

For more information on the sip.cfg file, refer to Application

Configuration on page A-4.

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

3 - 15

Administrator’s Guide SoundPoint IP / SoundStation IP

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 Create per-phone or per-platform <Ethernet address>.cfg files by

using the 00000000000.cfg and files from the distribution as templates.

For more information, refer to Master Configuration Files on page

A-2.

b 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.

c Edit the CONFIG_FILES attribute of the <Ethernet address>.cfg files

so that it references the appropriate sipXXXX.cfg file.

Note

For example, replace the reference to sip.cfg with sip650.cfg.

d Edit the LOG_FILE_DIRECTORY attribute of the <Ethernet

address>.cfg files so that it points to the log file directory.

e 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.

For more information, refer to Multiple Key Combinations on page C-9.

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.

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

For more information, refer to Parsing Vendor ID Information on page C-16.

3 - 16

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,

Supporting SoundPoint IP and SoundStation IP Phones.

Setting up Your System

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

SoundPoint IP 300 and 500 Phones on page 3-18.

Warning

The SoundPoint IP 300 and 500 phones will be supported on the latest
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.

3 - 17

Administrator’s Guide SoundPoint IP / SoundStation IP

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.

Warning

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.

Mandatory changes must be made or the software may not behave as expected.
For more information, refer to the “Configuration File Management on SoundPoint

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.

For more information, refer to Multiple Key Combinations on page C-9.

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.
Refer to Special Events <specialEvent/> on page A-15.

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/>
on page A-75.

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
model number. Refer to Master Configuration Files on page A-2.

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.

3 - 18

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.

3 - 19

Administrator’s Guide SoundPoint IP / SoundStation IP

Note

This approach takes advantage of an enhancement that was added in
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

http://www.polycom.com/support/voice/.

3 - 20

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:

• Setting Up Basic Features

• Setting Up Advanced Features

• Setting Up Audio Features

• Setting Up Security Features

4

This chapter also provides instructions on:

• Configuring SoundPoint IP / SoundStation IP Phones Locally

To troubleshoot any problems with your SoundPoint IP / SoundStation IP
phones on the network, refer to Troubleshooting Your SoundPoint IP /

SoundStation IP Phones on page 5-1. For more information on the

configuration files, refer to Configuration Files on page A-1.

Setting Up Basic Features

This section provides information for making configuration changes for the
following basic features:

• Call Log

• Call Timer

• Call Waiting

• Called Party Identification

• Calling Party Identification

• Missed Call Notification

4 - 1

Administrator’s Guide SoundPoint IP / SoundStation IP

• Connected Party Identification

• Context Sensitive Volume Control

• Customizable Audio Sound Effects

• Message Waiting Indication

• Distinctive Incoming Call Treatment

• Distinctive Ringing

• Distinctive Call Waiting

• Do Not Disturb

• Handset, Headset, and Speakerphone

• Local Contact Directory

• Local Digit Map

• Microphone Mute

• Soft Key Activated User Interface

• Speed Dial

• Time and Date Display

• Idle Display Animation

• Ethernet Switch

This section also provides information for making configuration changes for
the following basic call management features:

• Automatic Off-Hook Call Placement

• Call Hold

• Call Transfer

• Local / Centralized Conferencing

• Call Forward

• Directed Call Pick-Up

• Group Call Pick-Up

• Call Park/Retrieve

• Last Call Return

4 - 2

Call Log

Configuring Your System

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.

Central
(boot server)

Call Timer

Call Waiting

Note

On some SoundPoint IP platforms, missed calls and received calls appear in one
list. Missed 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:

Configuration File:

sip.cfg

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.

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.

and received calls appear as .

Enable or disable all call lists or individual call lists.

• For more information, refer to Feature <feature/> on page A-77.

Central
(boot server)

Configuration changes can performed centrally at the boot server:

Configuration File:

phone1.cfg

For related configuration changes, refer to Customizable Audio Sound Effects
on page 4-5.

Specify the ring tone heard on an incoming call when another call is
active.

• For more information, refer to Call Waiting <callWaiting/> on page

A-90.

4 - 3

Administrator’s Guide SoundPoint IP / SoundStation IP

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
(boot server)

Local Web Server

Configuration File:

sip.cfg

(if enabled)

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.

Note

On some SoundPoint IP platforms, missed calls and received calls appear in one
list.

Specify whether or not to use directory name substitution.

• For more information, refer to User Preferences <up/> on page

A-23.

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.

4 - 4

Configuring Your System

Configuration changes can performed centrally at the boot server:

Central
(boot server)

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

Turn this feature on or off.

• For more information, refer to Feature <feature/> on page A-77.

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.

<serverMissedCall/> on page A-89.

Configuration changes can performed centrally at the boot server:

Central
(boot server)

Configuration file:

sip.cfg

Adjust receive and handset/headset volume.

• For more information, refer to Volume Persistence <volume/> on

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)

page A-37.

4 - 5

Administrator’s Guide SoundPoint IP / SoundStation IP

Note

Note

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Local Web Server

Configuration File:

sip.cfg

(if enabled)

L16/16000 is not supported on SoundPoint IP 301 and SoundStation IP 4000
phones.

The alternate sampled audio sound effect files must be present on the boot server
or the Internet for downloading at boot time.

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

• For more information, refer to Sampled Audio for Sound Effects

<saf/> on page A-27 or Sound Effects <se/> on page A-28.

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).

• For more information, refer to User Preferences <up/> on page

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).

For related configuration changes, refer to Local Contact Directory on page

4-9.

4 - 6

on page A-97.

A-23.

Distinctive Ringing

Configuring Your System

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.

For more information, refer to Distinctive Incoming Call Treatment, the
previous section. This option has a higher priority than option 1 and a
lower priority than option 3.

3. The

voIpProt.SIP.alertInfo.x.class

specific ring types. This option has the highest priority.

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Local Local Phone User

Configuration file:

sip.cfg

Configuration file:

phone1.cfg
XML File: <Ethernet

address>-directory.
xml

Interface

voIpProt.SIP.alertInfo.x.value

and

fields can be used to map calls to

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

• For more information, refer to Alert Information <alertInfo/> on
page A-14.

Specify the ring type to be used for each line.

• For more information, refer to Registration <reg/> on page A-84.

This file can be created manually using an XML editor.

• For more information, refer to Local Contact Directory File Format
on page 4-10.

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.class

call waiting types, currently limited to two styles.

Configuration changes can performed centrally at the boot server:

Central
(boot server)

Configuration file:

sip.cfg

voIpProt.SIP.alertInfo.x.value

fields can be used to map calls to distinct

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

• For more information, refer to Alert Information <alertInfo/> on
page A-14.

and

4 - 7

Administrator’s Guide SoundPoint IP / SoundStation IP

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
on page 4-18.

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).

Note

Server-based DND is disabled if Shared Call Appearance or Bridged Line
Appearance is enabled.

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Local Local Phone User

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

Interface

Enable or disable server-based DND.

• For more information, refer to SIP <SIP/> on page A-10

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

• For more information, refer to Call Handling Configuration <call/>
on page A-55.

Enable or disable server-based DND as a per-registration feature.

• For more information, refer to Registration <reg/>on page A-84.

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

• For more information, refer to Do Not Disturb <dnd/> on page

A-93.

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.

4 - 8

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:

Configuring Your System

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

(if enabled)
Local Phone User

Interface

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.

Enable or disable persistent headset mode.

• For more information, refer to User Preferences <up/> on page

A-23.

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

Enable or disable persistent headset mode through the Settings
menu.

Changes are saved to local flash and backed up to <Ethernet
address>-phone.cfg on the boot server. Changes will permanently
override global settings unless deleted through the Reset Local
Config menu selection and the <Ethernet address>-phone.cfg is
removed from the boot server.

4 - 9

Administrator’s Guide SoundPoint IP / SoundStation IP

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Central
(boot server)
continued

Local Local Phone User

Configuration file:

sip.cfg

XML file:
000000000000-direct

ory.xml

XML file: <Ethernet
address>-directory.
xml

Interface

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

• For more information, refer to Directory <dir/> on page A-58.

A sample file named 000000000000-directory~.xml (Note the extra
“~” in the filename) is included with the application file distribution.
This file can be used as a template for the per-phone <Ethernet

address>-directory.xml directories (edit contents, then rename to
<Ethernet address>-directory.xml). It also can be used to seed

new phones with an initial directory (edit contents, then remove “~”
from file name). Telephones without a local directory, such as new
units from the factory, will download the 00000000000-directory.xml
directory and base their initial directory on it. These files should be
edited with an XML editor. These files can be downloaded once per
reflash.

For information on file format, refer to Local Contact Directory File

Format, the following section.

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.

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.

4 - 10

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>

...
<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

Configuring Your System

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 UTF-8 encoded string

last name

of up to 40 bytes

ct 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

divert contact
The forward-to address for the autodivert feature.

user part of a SIP
URL) or a string that
constitutes a valid SIP
URL

4 - 11

Administrator’s Guide SoundPoint IP / SoundStation IP

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 0,1 bu ddy watching

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

bb 0,1 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).

Central
(boot server)

Local Web Server

4 - 12

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

(if enabled)

Configuration changes can performed centrally at the boot server or locally:

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

• For more information, refer to Dial Plan <dialplan/> on page A-16.

Specify per-registration impossible match behavior, trailing #
behavior, digit map matching strings, and time out values that
override those in sip.cfg.

• For more information, refer to Dial Plan <dialplan/> on page A-93.

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.

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
information, refer to Presence on page 4-37.

Configuring Your System

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Local Local Phone User

XML file:

<Ethernet
address>-directory.
xml

Interface

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

The

<sd>x</sd>

file links a directory entry to a speed dial resource within the phone.
Speed dial entries are mapped automatically to unused line keys (line
keys are not available on the SoundStation IP 4000
are available for selection within the speed dial menu. (Press the
up-arrow key from the idle display to jump to SpeedDial).

• For more information, refer to Local Contact Directory File Format
on page 4-10.

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.

element in the <Ethernet address>-directory.xml

and 7000) and

4 - 13

Administrator’s Guide SoundPoint IP / SoundStation IP

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
(boot server)

Local Web Server

Configuration file:

sip.cfg

(if enabled)

Local Phone User
Interface

Turn time and date display on or off.

• For more information, refer to User Preferences <up/> on page

A-23.

Set the time and date display formats.

• For more information, refer to Date and Time <datetime/> on
page A-23.

Set the basic SNTP settings and daylight savings parameters.

• For more information, refer to Time Synchronization <sntp/> on
page A-51.

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.

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

For more information, refer to DHCP or Manual TCP/IP Setup on
page 3-2.

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
logo could be displayed (refer to Adding a Background Logo on page C-5).

4 - 14

Configuring Your System

Configuration changes can performed centrally at the boot server:

Central
(boot server)

Ethernet Switch

Configuration file:

sip.cfg

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.

To turn idle display animation on or off.

• For more information, refer to Indicators <ind/> on page A-65.

To replace the animation used for the idle display.

• For more information, refer to Animations <anim/> <IP_300/>,

<IP_330/>, <IP_400/>, <IP_500/>, <IP_600/>, <IP_4000/> on

page A-66.

To change the position of the idle display animation.

• For more information, refer to Graphic Icons <gi/> <IP_300/>,

<IP_330>, <IP_400/>, <IP_500/>, <IP_600/>, <IP_4000/> on

page A-68.

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

<QOS/> on page A-47.

4 - 15

Administrator’s Guide SoundPoint IP / SoundStation IP

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
(boot server)

Configuration file:

phone1.cfg

Call Hold

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

(if enabled)

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

• For more information, refer to Automatic Off-Hook Call Placement

<autoOffHook/> on page A-89.

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:

Specify whether RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or
a=inactive) outgoing hold signaling is used.

• For more information, refer to SIP <SIP/> on page A-10.

Specify local hold reminder options.

• For more information, refer to Hold, Local Reminder

<hold/><localReminder/> on page A-58.

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.

4 - 16

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).

Call Transfer

Configuring Your System

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
(boot server)

Configuration file:

sip.cfg

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

• For more information, refer to SIP <SIP/> on page A-10.

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.

Note

Conferences are not available when the G.729 codec is enabled on the
SoundStation IP 4000 conference phone.

on page A-55.

4 - 17

Administrator’s Guide SoundPoint IP / SoundStation IP

Configuration changes can performed centrally at the boot server:

Central
(boot server)

Call Forward

Configuration file:

sip.cfg

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

Specify the conference hold behavior (all parties on hold or only host
is on hold).

• For more information, refer to Call Handling Configuration <call/>
on page A-55.

Specify whether or not all parties hear sound effects while setting up
a conference.

• For more information, refer to Call Handling Configuration <call/>
on page A-55.

Specify which type of conference to establish and the address of the
centralized conference resource.

• For more information, refer to Conference Setup <conference/>
on page A-15.

4 - 18

• 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
(refer to Distinctive Incoming Call Treatment on page 4-6). The user’s ability
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).

Configuring Your System

Note

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

(if enabled)

Server-based call forwarding is disabled if Shared Call Appearance or Bridged Line
Appearance is enabled.

Enable or disable server-based call forwarding.

• For more information, refer to SIP <SIP/> on page A-10

Enable or disable server-based call forwarding as a per-registration
feature.

• For more information, refer to Registration <reg/>on page A-84

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.

• For more information, refer to Diversion <divert/> on page A-90.

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

Directed Call Pick-Up

Central
(boot server)

Configuration file:

sip.cfg

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.

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:

Turn this feature on or off.

• For more information, refer to Feature <feature/> on page A-77.

4 - 19

Administrator’s Guide SoundPoint IP / SoundStation IP

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
(boot server)

Configuration file:

sip.cfg

Call Park/Retrieve

Central
(boot server)

Configuration file:

sip.cfg

Last Call Return

Central
(boot server)

Configuration file:

sip.cfg

Turn this feature on or off.

• For more information, refer to Feature <feature/> on page A-77.

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:

Turn this feature on or off.

• For more information, refer to Feature <feature/> on page A-77.

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:

Turn this feature on or off.

• For more information, refer to Feature <feature/> on page A-77.

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

• For more information, refer to Call Handling Configuration <call/>
on page A-55.

Setting Up Advanced Features

This section provides information for making configuration changes for the
following advanced features:

• Configurable Feature Keys

• Multiple Line Keys per Registration

• Multiple Call Appearances

• Shared Call Appearances

4 - 20

Configuring Your System

• Bridged Line Appearance

• Busy Lamp Field

• Customizable Fonts and Indicators

• Instant Messaging

• Multilingual User Interface

• Downloadable Fonts

• Synthesized Call Progress Tones

• Microbrowser

• Real-Time Transport Protocol Ports

• Network Address Translation

This section also provides information for making configuration changes for
the following advanced call server features:

• Voice Mail Integration

• Multiple Registrations

• Automatic Call Distribution

• Server Redundancy

• Presence

• Microsoft Live Communications Server 2005 Integration

Configurable Feature Keys

All key functions can be changed from the factory defaults. The scrolling
timeout for specific keys can be configured.

Note

No feature keys on the SoundStation IP 4000 can be remapped.

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

— A speed-dial

the phones (Directories, Applications, Conference, Transfer, Redial,
Menu, Messages, Do Not Disturb, Call Lists)

— Null

4 - 21

Administrator’s Guide SoundPoint IP / SoundStation IP

• 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
(boot server)

Configuration File:

sip.cfg

Set the key scrolling timeout, key functions, and sub-pointers for each
key (usually not necessary).

• For more information, refer to Keys <key/> on page A-63.

For more information on the default feature key layouts, refer to Default

Feature Key Layouts on page C-10.

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
(boot server)

Local Web Server

Configuration file:

phone1.cfg

(if enabled)

Specify the number of line keys to assign per registration.

• For more information, refer to Registration <reg/> on page A-84.

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.

4 - 22

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.

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

Registration).

Configuration changes can performed centrally at the boot server or locally:

Configuring Your System

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

(if enabled)

Local Phone User
Interface

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

• For more information, refer to Call Handling Configuration <call/>
on page A-55.

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

• For more information, refer to Registration <reg/> on page A-84.

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.

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
information, refer to Shared Call Appearance Signaling on page B-10.

4 - 23

Administrator’s Guide SoundPoint IP / SoundStation IP

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

(if enabled)

Specify whether diversion should be disabled on shared lines.

• For more information, refer to Shared Calls <shared/> on page

A-57.

Specify line-seize subscription period.

• For more information, refer to Server <server/> on page A-7.

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

• For more information, refer to Special Events <specialEvent/> on
page A-15.

Specify per-registration line type (private or shared) and line-seize
subscription period if using per-registration servers. A shared line will
subscribe to a server providing call state information.

• For more information, refer to Registration <reg/> on page A-84.

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

• For more information, refer to Diversion <divert/> on page A-90.

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

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

4 - 24

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.

Configuring Your System

binds the appearances together logically and looks after the necessary state
notifications and performs an access control function. For more information,
refer to Bridged Line Appearance Signaling on page B-10.

Note

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

(if enabled)

In the configuration files, bridged lines are con fig ured by “shared line” parameters.

Specify whether diversion should be disabled on shared lines.

• For more information, refer to Call Handling Configuration <call/>
on page A-55.

Specify per-registration line type (private or shared) and the shared
line third party name. A shared line will subscribe to a server
providing call state information.

• For more information, refer to Registration <reg/> on page A-84.

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

• For more information, refer to Diversion <divert/> on page A-90.

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

Busy Lamp Field

Note

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.

This feature is available only on SoundPoint IP 600 phones and SoundPoint IP 601
and 650 phones with an attached Expansion Module.

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.

4 - 25

Administrator’s Guide SoundPoint IP / SoundStation IP

Polycom recommends that the BLF not be used in conjunction with the Microsoft
Live Communications Server 2005 feature. For more information, refer to Microsoft

Live Communications Server 2005 Integration on page 4-38.

Note

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

A-7).

Configuration changes can performed centrally at the boot server:

Central
(boot server)

Configuration file:

phone1.cfg

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

attendant.uri

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

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.

• For more information, refer to Fonts <font/> on page A-60.

Specify which bitmaps to use.

• For more information, refer to Bitmaps <bitmap/>on page A-65.

Specify how to create animations and LED indicator patterns.

• For more information, refer to Indicators <ind/> on page A-65.

.

A-98.

Instant Messaging

4 - 26

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

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

Configuring Your System

Note

Note

Note

This feature is not available on SoundPoint IP 301 phones.

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

Multilingual <ml/> on page A-20. (Note that within a Unicode range, some

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.

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
(boot server)

Local Local Phone User

Configuration file:

sip.cfg

Interface

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

• For more information, refer to Multilingual <ml/> on page A-20.
For instructions on adding new languages, refer to To add new

languages to those included with the distribution: on page A-21.

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.

4 - 27

Administrator’s Guide SoundPoint IP / SoundStation IP

Downloadable Fonts

New fonts can be loaded onto the phone. For guidelines on downloading
fonts, refer to Fonts <font/> on page A-60.

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
(boot server)

Microbrowser

Configuration file:

sip.cfg

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.

Note

As of SIP 2.2.0, the Services key and menu entry are renamed Applications,
however the functionality remains the same.

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

• For more information, refer to Chord-Sets <chord/> on page A-26.

Specify downloaded sampled audio files for advanced call progress
tones.

• For more information, refer to Sampled Audio for Sound Effects

<saf/> on page A-27.

Specify patterns.

• For more information, refer to Patterns <pat/> on page A-29 and

Call Progress Patterns on page A-30.

4 - 28

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.

Configuring Your System

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

(if enabled)

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

• For more information, refer to Microbrowser <mb/> on page A-79.

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.

4 - 29

Administrator’s Guide SoundPoint IP / SoundStation IP

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

(if enabled)

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:

Specify whether to filter incoming RTP packets by IP address,
whether to require symmetric port usage, whether to jam the
destination port and specify the local RTP port range start.

• For more information, refer to RTP <rtp/> on page A-49.

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.

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

(if enabled)

Voice Mail Integration

4 - 30

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

• For more information, refer to Network Address Translation

<nat/> on page A-97.

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.

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

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
(boot server)

Local Web Server

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

(if enabled)

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

• For more information, refer to User Preferences <up/> on page

A-23.

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.

• For more information, refer to Messaging <msg/> on page A-96.

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.

4 - 31

Administrator’s Guide SoundPoint IP / SoundStation IP

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

Local Web Server

(if enabled)

Specify the local SIP signaling port and an array of SIP servers to
register to. For each server specify the registration period and the
signaling failure behavior.

• For more information, refer to Local <local/> on page A-6 and

Server <server/> on page A-7.

For up to twelve registrations, specify a display name, a SIP address,
an optional display label, an authentication user ID and password, the
number of line keys to use, and an optional array of registration
servers. The authentication user ID and password are optional and
for security reasons can be omitted from the configuration files. The
local flash parameters will be used instead. The optional array of
servers and their associated parameters will override the servers
specified in sip.cfg if non-Null.

• For more information, refer to Registration <reg/> on page A-84.

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.

4 - 32

Configuring Your System

Local
(continued)

Local Phone User
Interface

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:

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.

For more information, refer to Local <local/> on page A-6, Server

<server/> on page A-7, and Registration <reg/> on page A-84.

Central
(boot server)

Central
(boot server)

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

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:

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

Turn this feature on or off.

• For more information, refer to Feature <feature/> on page A-77.

Enable this feature per registration.

• For more information, refer to Registration <reg/> on page A-84.

Turn this feature on or off.

• For more information, refer to Feature <feature/> on page A-77.

Enable this feature per registration.

• For more information, refer to Registration <reg/> on page A-84.

4 - 33

Administrator’s Guide SoundPoint IP / SoundStation IP

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.

Central
(boot server)

Note

Warning

Your SIP server provider should be consulted for recommended methods of
configuring phones and servers for fail-over configuration.

Prior to SIP 2.1, the
page A-84) could be used for fail-over configuration. The older behavior is no longer
supported. Customers that are using the
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

http://www.polycom.com/support/voice/.

Configuration changes can performed centrally at the boot server:

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

reg.x.server.y

Specify global primary and fallback server configuration parameters.

• For more information, refer to Protocol <volpProt/> on page A-6.

Specify per registration primary and fallback server configuration
parameters values that override those in sip.cfg.

• For more information, refer to Registration <reg/> on page A-84.

parameters (refer to Registration <reg/> on

reg.x.server.y

. configuration

4 - 34

Configuring Your System

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.

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

Note

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

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.
For more information, refer to Server <server/> on page A-7 and

Registration <reg/> on page A-84.

4 - 35

Administrator’s Guide SoundPoint IP / SoundStation IP

`

`

`

`

PSTN Gateway

Internet

PSTN

Hosted VoIP Service

Provider

VoIP SMB Customer

Premise

DNS Server

Call Server 1A

Call Server 1B

SIP Capable Router

Server2

Warning

If DNS is used to resolve the address for Servers, the DNS server is unavailable,
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.

4 - 36

Note

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

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.register

However, the INVITE will fail over to that server if all higher priority servers are
down.

is set to 0, then phone will not register to that server.

Presence

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.

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

4 - 37

Administrator’s Guide SoundPoint IP / SoundStation IP

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.

Note

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

The presence feature works differently when Microsoft Live Communications
Server 2005 is used as the call server. For more information, refer to the
following section, Microsoft Live Communications Server 2005 Integration.

Configuration changes can performed centrally at the boot server:

Central
(boot server)

Local Local Phone User

XML file: <Ethernet
address>-directory.
xml

Interface

The <bw>0</bw> (buddy watching) and <bb>0</bb> (buddy
blocking) elements in the <Ethernet address>-directory.xml file
dictate the Presence aspects of directory entries.

• For more information, refer to Local Contact Directory File Format
on page 4-10.

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

4 - 38

Note

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.

Polycom recommends that the BLF not be used in conjunction with the Microsoft
Live Communications Server 2005 feature. For more information, refer to Busy

Lamp Field on page 4-25.

Configuring Your System

Configuration changes can performed centrally at the boot server:

Central
(boot server)

Configuration file:

sip.cfg

Configuration file:

phone1.cfg

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.

Specify that support for Microsoft Live Communications Server 2005
is enabled.

• For more information, refer to SIP <SIP/> on page A-10.

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

• For more information, refer to Presence <pres/> on page A-60.

Turn the presence and messaging features on or off.

• For more information, refer to Feature <feature/> on page A-77.

Specify the number of line keys to assign per registration.

• For more information, refer to Registration <reg/> on page A-84.

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

• For more information, refer to Roaming Buddies

<roaming_buddies/> on page A-99.

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

• For more information, refer to Roaming Privacy

<roaming_privacy/> on page A-99.

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 Open sip.cfg in an XML editor.

b Locate the feature parameter.

c For the

feature.1.enabled

d For the

feature.2.enabled

feature.1.name = presence

attribute, set

to 1.

feature.2.name = messaging

to 1.

attribute, set

4 - 39

Administrator’s Guide SoundPoint IP / SoundStation IP

Note

e Locate the

Set the

voIpProt

parameter.

voIpProt.server.x.transport

attribute to TCPpreferred or

TLS.

Your selection depends on the LCS configuration.

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

f Set the

For example,

g Set the

h (Optional) If SIP forking is desired, set

voIpProt.server.x.address

to the LCS address.

voIpProt.server.1.address = "lcs2005.local"

voIpProt.SIP.lcs

attribute to 1.

voIpProt.SIP.ms-forking

attribute to 1.

Refer to SIP <SIP/> on page A-10.

i Save the modified sip.cfg configuration file.

2. Modify the phone1.cfg configuration file as follows:

a Open phone1.cfg in an XML editor.

b Locate the registration parameter.

c Set the

reg.1.address

to the LCS address.

For example,

d Set the

e (Optional) Set the

reg.1.address = "7778"

reg.1.server.y.address

reg.1.server.y.transport

to the LCS server name.

attribute to

TCPpreferred or TLS.

Your selection depends on the LCS configuration.

f Set

g Set

h Locate the

i Set the

reg.1.auth.userId

For example,

reg.1.auth.userId = "jbloggs"

reg.1.auth.password

For example,

reg.1.auth.password = "Password2"

roaming_buddies

roaming_buddies.reg

to the phone's LCS username.

to the LCS password.

attribute.

element to 1.

Refer to Roaming Buddies <roaming_buddies/> on page A-99.

j Locate the

k Set the

roaming_privacy

attribute.

roaming_privacy.reg

element to 1.

Refer to Roaming Privacy <roaming_privacy/> on page A-99.

l Save the modified phone1.cfg configuration file.

4 - 40

Configuring Your System

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 Open sip.cfg in an XML editor.

b Locate the feature parameter.

c For the

feature.1.enabled

d For the

feature.2.enabled

e Locate the

f If SIP forking is desired, set

feature.1.name = presence

attribute, set

to 1.

feature.2.name = messaging

to 1.

voIpProt

parameter.

voIpProt.SIP.ms-forking

attribute, set

attribute to 1.

Refer to SIP <SIP/> on page A-10.

g Save the modified sip.cfg configuration file.

2. Modify the phone1.cfg configuration file as follows:

a Open phone1.cfg in an XML editor.

b Locate the registration parameter.

c Select a registration to be used for the Microsoft Live Communications

Server 2005.

Typically, this would be 2.

d Set the

For example,

e Set the

f (Optional) Set the

reg.x.address

to the LCS address.

reg.2.address = "7778"

reg.x.server.y.address

reg.2.server.y.transport

to the LCS server name.

attribute to

TCPpreferred or TLS.

Your selection depends on the LCS configuration.

g Set

h Set

i Locate the

j Set the

reg.x.auth.userId

For example,

reg.2.auth.userId = "jbloggs"

reg.x.auth.password

For example,

reg.2.auth.password = "Password2"

roaming_buddies

roaming_buddies.reg

to the phone's LCS username.

to the LCS password.

attribute.

element to the number corresponding

to the LCS registration.

For example,

roaming_buddies.reg = 2

Refer to Roaming Buddies <roaming_buddies/> on page A-99.

k Locate the

roaming_privacy

attribute.

4 - 41

Administrator’s Guide SoundPoint IP / SoundStation IP

l Set the

to the LCS registration.

For example,

Refer to Roaming Privacy <roaming_privacy/> on page A-99.

m Save the modified phone1.cfg configuration file.

roaming_privacy.reg

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

• Jitter Buffer and Packet Error Concealment

• Voice Activity Detection

• DTMF Tone Generation

• DTMF Event RTP Payload

element to the number corresponding

roaming_privacy.reg = 2

• Acoustic Echo Cancellation

• Audio Codecs

• Background Noise Suppression

• Comfort Noise Fill

• Automatic Gain Control

• IP Type-of-Service

• IEEE 802.1p/Q

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

4 - 42

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
(boot server)

Local Web Server

Configuration file:

sip.cfg

(if enabled)

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.

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

• For more information, refer to Codec Profiles <audioProfile/> on
page A-36.

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.

Configuration changes can performed centrally at the boot server:

Central
(boot server)

Configuration file:

sip.cfg

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

Enable or disable VAD and set the detection threshold.

• For more information, refer to Voice Activity Detection <vad/> on
page A-47.

4 - 43

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
(boot server)

Configuration file:

sip.cfg

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
(boot server)

Configuration file:

sip.cfg

Acoustic Echo Cancellation

Set the DTMF tone levels, autodialing on and off times, and other
parameters.

• For more information, refer to Dual Tone Multi-Frequency

<DTMF/> on page A-25.

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

• For more information, refer to Dual Tone Multi-Frequency

<DTMF/> on page A-25.

4 - 44

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

For more information, refer to Acoustic Echo Cancellation <aec/> on page

A-34, Acoustic Echo Suppression <aes/> on page A-41, and Background
Noise Suppression <ns/> on page A-42.

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.

Note

AEC on the SoundPoint IP 301 handset is not supported.

Audio Codecs

The following table summarizes the phone’s audio codec support:

Effective

Sample

Algorithm MIME Type Ref. Bit Rate

G.711μ-law PMCU RFC 1890 64 Kbps 8 Ksps 10ms - 80ms 3.5KHz
G.711a-law PCMA RFC 1890 64 Kbps 8 Ksps 10ms - 80ms 3.5KHz
G.722 G722/8000 RFC 1890 64 Kbps 16 Ksps 10ms - 80ms 7 KHz
G.722.1 G722/16000 RFC 3047 16 Kbps,

24 Kbps,

32 Kbps
G.729AB G729 RFC 1890 8 Kbps 8 Ksps 10ms - 80ms 3.5KHz
SID CN RFC 3389 N/A N/A N/A N/A
RFC 2833 phone-event RFC 2833 N/A N/A N/A N/A

Rate Frame Size

16 Ksps 20ms - 80ms 7 KHz

audio
bandwidth

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

(if enabled)

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

• For more information, refer to Codec Preferences <codecPref/>
on page A-35 and Codec Profiles <audioProfile/> on page A-36.

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.

4 - 45

Administrator’s Guide SoundPoint IP / SoundStation IP

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

IP Type-of-Service

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

(if enabled)

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.

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:

Specify protocol-specific IP TOS settings.

• For more information, refer to IP TOS <IP/> on page A-48.

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

4 - 46

IEEE 802.1p/Q

Configuring Your System

The phone will tag all Ethernet packets it transmits with an 802.1Q VLAN
header for one of the following reasons:

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

• When it is instructed to tag packets through Cisco Discovery Protocol

(CDP) running on a connected Ethernet switch

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

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
(boot server)

Local Web Server

Configuration file:

sip.cfg

(if enabled)
Local Phone User

Interface

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

• For more information, refer to Ethernet IEEE 802.1p/Q

<ethernet/> on page A-47.

Specify 802.1p/Q settings.
Navigate to http://<phoneIPAddress>/netConf.htm#qo

Specify whether CDP is to be used or manually set the VLAN ID or
configure DHCP VLAN Discovery.

Phase 1: bootRom - Navigate to: SETUP menu during auto-boot
countdown.

Phase 2: Application - Navigate to:
Menu>Settings>Advanced>Admin Settings>Network Configuration

• For more information, refer to Setting Up the Network on page

3-2.

Setting Up Security Features

This section provides information for making configuration changes for the
following security-related features:

• Local User and Administrator Privilege Levels

• Custom Certificates

• Incoming Signaling Validation

• Configuration File Encryption

4 - 47

Administrator’s Guide SoundPoint IP / SoundStation IP

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
(refer to Configuring SoundPoint IP / SoundStation IP Phones Locally on page

4-50).

Configuration changes can performed centrally at the boot server or locally:

Central
(boot server)

Local Web Server

Configuration file:

sip.cfg

(if enabled)
Local Phone User

Interface

Custom Certificates

Specify the minimum lengths for the user and administrator
passwords.

• For more information, refer to Password Lengths

<pwd/><length/> on page A-74.

None.

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

Parameter Configuration on page A-100). Passwords can consist of

ASCII characters 32-127 (0x20-0x7F) only.
Changes are saved to local flash but are not backed up to <Ethernet

address>-phone.cfg on the boot server for security reasons.

The phone trusts certificates issued by widely recognized certificate
authorities when trying to establish a connection to a boot server for
application provisioning. Refer to Trusted Certificate Authority List on page

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.

Note

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/ .

Configuration changes can performed locally:

Local Local Phone User

Interface

4 - 48

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

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:

Configuring Your System

Central
(boot server)

Configuration File:

sip.cfg

Configuration File Encryption

Configuration files (excluding the master configuration file), contact
directories, and configuration override files can all be encrypted.

Note

The SoundPoint IP 300 and 500 phones will always fail at decrypting files. These
phones will recognize that a file is encrypted, but cannot decrypt it and will display
an error. Encrypted configuration files can only be decrypted on the SoundPoint IP
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
information, refer to Master Configuration Files on page A-2.

For more information on encrypting configuration files including determining
whether an encrypted file is the same as an unencrypted file and using the
SDK to facilitate key generation, refer to Encrypting Configuration Files on
page C-3.

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

• For more information, refer to Request Validation

<requestValidation/> on page A-14.

Central
(boot server)

Configuration changes can performed centrally at the boot server:

Configuration File:

sip.cfg

Configuration file:

<device>.cfg

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

• For more information, refer to Encryption <encryption/>
on page A-74.

Change the encryption key.

• For more information, refer to refer to Flash Parameter

Configuration on page A-100.

4 - 49

Administrator’s Guide SoundPoint IP / SoundStation IP

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
provisioning model. For more information, refer to Web Server <httpd/> on
page A-54.

The phone’s local user interface also permits many application settings to be
modified, such as SIP server address, ring type, or regional settings such as
time/date format and language.

Local Web
Server Access

Local Settings
Menu Access

Passwords:

Administrator
password
required.

User password
required.

Point your web browser to http://<phoneIPAddress>/.
Configuration pages are accessible from the menu along the top banner.
The web server will issue an authentication challenge to all pages except for

the home page.
Credentials are (case sensitive):
User Name: Polycom
Password: The administrator password is used for this.

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

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

Restart Phone

4 - 50

Warning

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 Rese t Local Config menu selection.

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:

• BootROM Error Messages

• Application Error Messages

• Status Menu

• Log Files

This chapter also presents phone issues, likely causes, and corrective actions.
Issues are grouped as follows:

• Power and Startup

• Controls

• Access to Screens and Systems

• Calling

• Displays

• Audio

• Upgrading

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

http://www.polycom.com/support/voice/.

If your problems is not listed in this chapter nor described in the latest Release
Notes, contact your Certified Polycom Reseller for support.

5 - 1

Administrator’s Guide SoundPoint IP / SoundStation IP

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.

5 - 2