Poly SoundPoint IP 550 SoundPoint IP, SoundStation IP and Polycom VVX Administrator’s Guide, Version 3.3.0

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Administrator’s Guide for the Polycom® UC Software

3.3.0 | June 2010 | 1725-11530-330 Rev. A

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Trademark Information

POLYCOM®, the Polycom “Triangles” logo and the names and marks associated with Polycom’s products are trademarks and/or service marks of Polycom, Inc. and are registered and/or common law marks in the United States and various other countries. All other trademarks are property of their respective owners. No portion hereof may be reproduced or transmitted in any form or by any means, for any purpose other than the recipient’s personal use, without the express written permission of Polycom.

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.

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 © 1998-2008 The OpenSSL Project Copyright © 1995-1998 Eric Young (eay@cryptsoft.com). All rights reserved Copyright © 1995-2002 Jean-Loup Gailly and Mark Adler Copyright © 1996-2008, Daniel Stenberg, <daniel@haxx.se>

Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the “Software”), to deal in the Software without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so, subject to the following conditions:

The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software.

THE SOFTWARE IS PROVIDED “AS IS”, WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.

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.

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About This Guide

The Administrator’s Guide for the Polycom® UC Software is for administrators who need to configure, customize, manage, and troubleshoot Polycom® SoundPoint® IP, SoundStation® IP, and VVX® phones. This guide covers the SoundPoint IP 320, 321, 330, 331, 335, 450, 550, 560, 650, and 670 desktop phones, the SoundStation IP 5000, 6000 and 7000 conference phones, and the VVX 1500 business media phone.

The following related documents for the SoundPoint IP, SoundStation IP, and VVX phones 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

• Web Applications Developer’s Guide, which assists in the development of applications that run on the SoundPoint IP and SoundStation IP phone’s Microbrowser and the VVX 1500 phone’s Browser

• Technical Bulletins, which describe workarounds to existing issues and provide expanded descriptions and examples

• 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/.

Polycom recommends that you record the phone model numbers, software (both the BootROM and UC Software), and partner platform for future reference.

SoundPoint IP, SoundStation IP, and VVX models: _____________________

BootROM version: ________________________________________________

UC Software version: ______________________________________________

Partner Platform: _________________________________________________

reseller or go to Polycom

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Administrator’s Guide for the Polycom UC Software

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

1 Introducing the Polycom UC Software Family of Phones . . . 1–1

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

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

SoundStation IP Conference Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1–4

VVX 1500 Business Media Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1–5

Key Features of Your Polycom Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1–6

Where Polycom Phones Fit . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–2

Polycom Phone Software Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–3

BootROM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–4

Polycom UC Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–5

Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–6

Resource Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–9

Available Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–9

New Features in Polycom UC Software 3.3.0 . . . . . . . . . . . . . . . . . . . . . . 2–16

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 . . . . . . . . . . . . . . . . . . . . . . . . . 3–6

Setting Up the Provisioning Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–14

Deploying Phones From the Provisioning Server . . . . . . . . . . . . . . . . . . . 3–17

Upgrading Polycom UC Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–20

Supporting Current SoundPoint IP, SoundStation IP, and VVX

Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–21

Supporting Legacy SoundPoint IP and SoundStation IP Phones . . 3–22

Provisioning SoundStation IP 7000 Phones Using C-Link . . . . . . . . 3–24

Provisioning VVX 1500 Phones Using a Polycom CMA System . . . 3–25

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Administrator’s Guide for the Polycom UC Software

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

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

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

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

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

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

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

Missed Call Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–5

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

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

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

Distinctive Ringing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–6

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

Do Not Disturb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–7

Handset, Headset, and Speakerphone . . . . . . . . . . . . . . . . . . . . . . . . . 4–8

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

Local Digit Map . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–13

Microphone Mute . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–15

Soft Key Activated User Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–15

Speed Dial . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–15

Time and Date Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–16

Idle Display Image Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–17

Ethernet Switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–18

Graphic Display Backgrounds . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–18

Automatic Off-Hook Call Placement . . . . . . . . . . . . . . . . . . . . . . . . . . 4–20

Call Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–20

Call Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–21

Local / Centralized Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–21

Call Forward . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–22

Directed Call Pick-Up . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–24

Group Call Pick-Up . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–25

Call Park/Retrieve . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–25

Last Call Return . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–26

Setting Up Advanced Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–26

Configurable Feature Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–27

Multiple Line Keys per Registration . . . . . . . . . . . . . . . . . . . . . . . . . . 4–28

Multiple Call Appearances . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–29

Customizable Fonts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–30

Instant Messaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–30

Multilingual User Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–31

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Contents

Downloadable Fonts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–32

Synthesized Call Progress Tones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–32

Browser and Microbrowser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–33

Real-Time Transport Protocol Ports . . . . . . . . . . . . . . . . . . . . . . . . . . 4–34

Network Address Translation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–35

Corporate Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–35

CMA Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–38

Recording and Playback of Audio Calls . . . . . . . . . . . . . . . . . . . . . . . 4–38

Digital Picture Frame . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–39

Enhanced Feature Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–40

Configurable Soft Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–45

LCD Power Saving . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–48

Shared Call Appearances . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–48

Bridged Line Appearance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–50

Busy Lamp Field . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–51

Voice Mail Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–52

Multiple Registrations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–53

SIP-B Automatic Call Distribution . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–55

Feature Synchronized Automatic Call Distribution . . . . . . . . . . . . . 4–55

Server Redundancy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–56

Presence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–60

CMA Presence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–61

Microsoft Live Communications Server 2005 Integration . . . . . . . . 4–61

Access URL in SIP Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–64

Static DNS Cache . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–68

Display of Warnings from SIP Headers . . . . . . . . . . . . . . . . . . . . . . . 4–72

Quick Setup of Polycom Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–73

Setting Up Audio Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–73

Customizable Audio Sound Effects . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–74

Context Sensitive Volume Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–75

Low-Delay Audio Packet Transmission . . . . . . . . . . . . . . . . . . . . . . . 4–75

Jitter Buffer and Packet Error Concealment . . . . . . . . . . . . . . . . . . . . 4–75

Voice Activity Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–76

DTMF Tone Generation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–76

DTMF Event RTP Payload . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–76

Acoustic Echo Cancellation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–77

Audio Codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–77

Background Noise Suppression . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–81

Comfort Noise Fill . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–81

Automatic Gain Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–81

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Administrator’s Guide for the Polycom UC Software

IP Type-of-Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–82

IEEE 802.1p/Q . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–82

Voice Quality Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–83

Dynamic Noise Reduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–84

Treble/Bass Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–84

Audible Ringer Location . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–85

Setting Up Video Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–85

Video Transmission . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–85

Video Codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–86

H.323 Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–87

Setting Up Security Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–91

Local User and Administrator Privilege Levels . . . . . . . . . . . . . . . . . 4–92

Custom Certificates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–92

Incoming Signaling Validation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–93

Secure Real-Time Transport Protocol . . . . . . . . . . . . . . . . . . . . . . . . . 4–93

Configuration File Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–94

Digital Certificates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–95

Mutual TLS Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–97

Secure Real-Time Transport Protocol . . . . . . . . . . . . . . . . . . . . . . . . . 4–98

Configurable TLS Cipher Suites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–100

Locking the Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–101

Support for EAPOL Logoff Message . . . . . . . . . . . . . . . . . . . . . . . . . 4–102

Configuring Polycom Phones Locally . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–103

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5 Troubleshooting Your Polycom Phones . . . . . . . . . . . . . . . . 5–1

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

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

Polycom UC Software Error Messages . . . . . . . . . . . . . . . . . . . . . . . . . 5–3

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

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

Reading a Boot Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–8

Reading an Application Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–9

Reading a Syslog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–10

Testing Phone Hardware . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–10

Uploading Phone’s Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–11

Power and Startup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–12

Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–13

Access to Screens and Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–14

Calling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–15

Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–16

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Contents

Audio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–17

Licensable Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–17

Upgrading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–18

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

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

Sample Template Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–6

Configuration Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–8

<acd/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–10

<apps/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–10

<attendant/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–13

<bg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–16

<bitmap/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–20

<call/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–21

<device/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–30

<dialplan/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–34

<dir/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–43

<divert/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–48

<dns/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–51

<efk/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–53

<feature/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–58

<font/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–61

<httpd/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–63

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

<lcl/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–65

<license/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–68

<mb/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–69

<msg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–72

<nat/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–73

<phoneLock/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–74

<pnet/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–76

<powerSaving/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–76

<pres/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–78

<prov/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–78

<qos/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–79

<reg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–82

<request/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–92

<roaming_buddies/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–92

<roaming_privacy/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–93

<saf/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–93

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Administrator’s Guide for the Polycom UC Software

<se/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–95

<sec/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–101

<softkey/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–108

<tcpIpApp/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–112

<tones/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–118

<up/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–120

<video/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–125

<voice/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–134

<voIpProt/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–141

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–10

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–4

Changing the Key on the Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–6

Adding a Customizable Logo on the Idle Display . . . . . . . . . . . . . . . . . . C–6

BootROM/SIP Software Dependencies . . . . . . . . . . . . . . . . . . . . . . . . . . . C–8

Migration Dependencies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–9

Supported SoundStation IP 7000 / Polycom HDX Software

Interoperability . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–9

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

Default Feature Key Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–11

Internal Key Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–17

Assigning a VLAN ID Using DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–21

Parsing Vendor ID Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–22

Product, Model, and Part Number Mapping . . . . . . . . . . . . . . . . . . . . . C–24

Disabling PC Ethernet Port . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–25

Modifying Phone’s Configuration Using the Web Interface . . . . . . . . . C–25

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Contents

Capturing Phone’s Current Screen . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–28

LLDP and Supported TLVs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–28

Supported TLVs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–30

D Technical Support Configuration Parameters . . . . . . . . . . .D–1

<device/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D–1

<key/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D–3

<lcl/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D–3

<log/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D–4

<voice/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D–7

E Third Party Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . E–1

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

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Administrator’s Guide for the Polycom UC Software

xii

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Introducing the Polycom UC Software Family of Phones

This chapter introduces the family of Polycom® phones that run the Polycom® UC Software, which is described in this guide.

This family of phones 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. These phones support advanced functionality, including multiple call and flexible line appearances, HTTPS secure provisioning, presence, custom ring tones, and local conferencing.

These phones are endpoints 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

• SoundStation IP Conference Phones

• VVX 1500 Business Media Phone

For a list of key features available on these phones running the latest software, refer to Key Features of Your Polycom Phones on page 1-6.

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/voicedocumentation/. Additional options are

also available. For more information, contact your Polycom distributor.

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Note

Documentation for the SoundPoint IP 300, 301, 430, 500, 501, 600, and 601 desktop phones is available at http://www.polycom.com/voicedocumentation/ . These ‘legacy’ phones are not directly supported by the newest software, Polycom UC Software 3.3.0 .

The currently supported desktop phones are:

• SoundPoint IP 320/321/330/331/335

• SoundPoint IP 450

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• SoundPoint IP 550/560

• SoundPoint IP 650

Introducing the Polycom UC Software Family of Phones

• SoundPoint IP 670

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SoundStation IP Conference Phones

This section describes the current SoundStation® IP conference phones. For individual guides, refer to the product literature available at

http://www.polycom.com/voicedocumentation/. Additional options are

also available. For more information, contact your Polycom distributor.

Note

Documentation for the SoundStation IP 4000 conference phone is available at

http://www.polycom.com/voicedocumentation/ . This ‘legacy’ phone is not directly

supported by the newest software, Polycom UC Software 3.3.0 .

The currently supported conference phones are:

• SoundStation IP 5000

• SoundStation IP 6000

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• SoundStation IP 7000

VVX 1500 Business Media Phone

Introducing the Polycom UC Software Family of Phones

This section describes the current VVX® 1500 business media phone. For the individual guide, refer to the product literature available at

http://www.polycom.com/voicedocumentation/. Additional options are

also available. For more information, contact your Polycom distributor.

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Administrator’s Guide for the Polycom UC Software

Key Features of Your Polycom Phones

The key features of the Polycom phones running Polycom UC Software are:

• Award winning sound quality and full-duplex speakerphone or conference phone

— Permits natural, high-quality, two-way conversations

— Uses Polycom’s industry leading Acoustic Clarity Technology

— Most phone models support Polycom’s HDVoice™ technology

• Easy-to-use

— An easy transition from traditional PBX systems into the world of IP

Communications

— 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 phones as standards develop and protocols evolve

— Extends the life of the phone to protect your investment

— 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

— 102 x 23 pixel graphical LCD for the SoundPoint IP

320/321/330/331/335

— 256 x 116 pixel graphical grayscale LCD for the SoundPoint IP 450

(supports Asian characters)

— 320 x 160 pixel graphical grayscale LCD for the SoundPoint IP

550/560/650 (supports Asian characters)

— 320 x 160 pixel graphical color LCD for the SoundPoint IP 670

(supports Asian characters)

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— 248x 68 pixel graphical LCD for SoundStation IP 5000

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Introducing the Polycom UC Software Family of Phones

— 248 x 68 pixel graphical LCD for the SoundStation IP 6000

— 256 x 128 pixel graphical grayscale LCD for the SoundStation IP 7000

— 800 x 480 pixel graphical color LCD for the VVX 1500 (touch screen)

• Dual auto-sensing 10/100/1000baseT Ethernet ports

— Leverages existing infrastructure investment

— No re-wiring with existing CAT 5 cabling

— Simplifies installation

— 1000baseT is supported by the SoundPoint IP 560 and 670 and VVX

1500 only

• Power over Ethernet (PoE) port or Power pack option

— Built-in IEEE 802.3af PoE port on the SoundPoint IP

320/321/330/331/335, 450, 550, 560, 650, and 670, the SoundStation IP 5000, 6000 and 7000, and VVX 1500 (auto-sensing)

— Unused pairs on Ethernet port are used to deliver power to the phone

via a wall adapter allowing fewer wires to desktop (for the SoundStation IP 6000 and 7000 conference phones)

• Multiple language support on most phones

— Set on-screen language to your preference. Select from

Chinese (Simplified), Danish, Dutch, English (Canada, United Kingdom, and United States), French, German, Italian, Japanese, Korean, Norwegian, Polish, Portuguese (Brazilian), Russian, Slovenian, Spanish (International), and Swedish.

— Chinese (Simplified), Japanese, and Korean are not supported on the

SoundPoint IP 320/321/330/331/335 phones.

• Microbrowser

— Supports a subset of XHTML constructs; otherwise runs like any other

Web browser.

• Browser on the Polycom VVX 1500

— Supports XHTML 1.1 constructs, HTML 4.01, JavaScript, CCS 2.1, and

SVG 1.1 (partial support).

• XML status/control API

— Ability to poll phones for call status and device information.

— Ability to receive telephony notification events.

For more information, refer to Web Applications Developer’s Guide, which is available at http://www.polycom.com/voicedocumentation/

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Page 21

Overview

This chapter provides an overview of the Polycom® UC Software and how the phones fit into the network configuration.

The UC Software supports the deployment of Polycom phones in several deployment scenarios:

• As a SIP based endpoint interoperating with a SIP call server or soft-switch. For more information, review the remainder of this chapter.

• As an H.323 video endpoint (Polycom VVX® 1500). For more information, on using phones in a strict H.323 environment, refer to the Deployment Guide for the Polycom® VVX® 1500 D Business Media Phone, which is available from

http://www.polycom.com/support/video/business_media_phones/vv x1500d.html .

• In conjunction with a Polycom HDX video system (SoundStation® IP

7000) . For more information on using phones with a Polycom HDX system, refer to Integration Guide for the Polycom® SoundStation® IP 7000 Conference Phone Connected to a Polycom® HDX® System, which is available from

http://www.polycom.com/support/voice/soundstation_ip_series/sou ndstation_ip7000_hdx_series.html .

SIP is the Internet Engineering Task Force (IETF) standard for multimedia communications 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 Polycom phones to successfully operate as a SIP endpoint in your network, they must meet the following requirements:

• A working IP network is established.

• Routers are configured for VoIP.

• VoIP gateways are configured for SIP.

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• The latest (or compatible) Polycom UC Software 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 Polycom Phones Fit

• Polycom Phone Software Architecture

• Available Features

• New Features in Polycom UC Software 3.3.0

To set up your Polycom phones on the network, refer to Setting up Your

System on page 3-1. To configure your Polycom phones with the desired

features, refer to Configuring Your System on page 4-1. To troubleshoot any problems with your Polycom phones on the network, refer to Troubleshooting

Your Polycom Phones on page 5-1.

Where Polycom Phones Fit

The phones connect physically to a standard office twisted-pair (IEEE 802.3) 10/100/1000 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. As

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Overview

Local Application

Server

Local

Boot Server

10/100/1000

Ethernet

Switch

Voice Bridge

Ethernet Switches

Router/ Firewall

PCPC

10/100/1000

Ethernet

Hub

Internet

PSTN

Remote

Boot Server

Remote

Application

Server

PSTN Gateway

Polycom phones run the same protocols as your office personal computer, many innovative applications can be developed without resorting to specialized technology.

Polycom Phone Software Architecture

The architecture of the Polycom phone software is made of 4 basic components:

• BootROM—loads first when the phone is powered on

• Polycom UC Software—software that implements the phone and the

related functionality of the device

• Configuration—configuration parameters stored in separate files

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Administrator’s Guide for the Polycom UC Software

• Resource Files—optional, needed by some of the advanced features

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

and 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 provisioning server (or boot server) to look for any updates to the BootROM application.

If updates are found, they are downloaded and saved to flash memory, eventually overwriting itself after verifying the integrity of the download.

4. If a new BootROM is downloaded, formats the file system clearing out any application software and configuration files that may have been present.

5. Downloads the master configuration file.

This file is either called <MAC-address>.cfg or 000000000000.cfg . This file is used by the BootROM and the application for a list of other files that are needed for the operation of the phone.

6. Examines the master configuration file for the name of the application file, and then looks for this file on the provisioning server.

If the copy on the provisioning server is different than the one stored in flash memory or there is no file stored in flash memory, the application file is downloaded.

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Polycom UC Software

Overview

7. Extracts the application from flash memory.

8. Installs the application into RAM, then uploads a log file with events

from the boot cycle.

The BootROM will then terminate, and the application takes over.

The Polycom UC Software manages the VoIP stack, the digital signal processor (DSP), the user interface, and the network interaction. UC Software manages everything to do with the phone’s operation. UC software implements the following functions and features:

• VoIP signalling for a wide range of voice and video telephony functions using SIP signalling for call setup and control.

• H.323 signalling for video telephony.

• Industry standard security techniques for ensuring that all provisioning,

signalling, and media transactions are robustly authenticated and encrypted.

• Advanced audio signal processing for handset, headset, and speakerphone communications using a wide range of audio codecs.

• Flexible provisioning methods to support single phone, small business, and large multi-site enterprise deployments.

The software is a single file binary image and contains a digital signature to prevent tampering or loading rogue software images.

There is a new image file in each release of software.

The software performs the following tasks in order:

1. Downloads system, per-phone configuration, and resource files.

There are a number of configuration template files (for example, sip-basic.cfg and reg-basic.cfg). Customize these files for your own use. Include only those that you need. For more information, refer to Sample

Template Files on page A-6.

2. Controls all aspects of the phone.

3. Uploads log files.

BootROM and Polycom UC Software Wrapper

Both the BootROM and Polycom UC Software run on multiple platforms (meaning all previously released versions of hardware that are still supported).

Current build archives have both split and combined images, so it is up to the administrator which model(s) to support. Using split files saves a lot of internal network traffic during reboots and updates.

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Configuration

Note

Warning

Note

As of SIP 3.2.2, the VVX 1500 BootROM and SIP software were distributed as a one package.

The Polycom phones can be configured automatically through files stored on a central provisioning server, manually through the phone’s local UI or web interface, or by using a combination of the automatic and manual methods.

The recommended method for configuring phones is automatically through a central provisioning server, but if one is not available, the manual method will allow changes to most of the key settings.

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 UC Software must be used together with that software. Failure to do this may render the phone unusable.

You can make changes to the configuration files through the web interface to the phone. Using your chosen browser, enter the phone’s IP address as the browser address. For more information, refer to Modifying Phone’s Configuration Using the

Web Interface on page C-25.

Changes made through the web interface are written to the override file (highest priority). These changes remain active and will take precedence over the configuration files stored on the provisioning server until Reset Web Configuration is performed.

The precedence order for configuration parameter changes is as follows (highest to lowest):

• User changes through the phone’s user interface

• Web configuration through a browser

• Polycom CMA system

• Configuration files

• Default values

2 - 6

The phone configuration files consist of:

• Master Configuration Files

• Polycom UC Software Configuration Files

• Override Files

This section also contains information on:

• Central Provisioning

• Manual Configuration

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Overview

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.

Polycom UC Software Configuration Files

Configuration files can be arranged in a flexible manner and 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. Configuration parameters may be included in more than one configuration file. In this case, the parameter encountered first when reading the configuration files from left to right in the <MACAddress>.cfg file will take precedence. As of Polycom UC Software 3.3.0, the use of configuration files is optional, which means that the phone will attempt to boot up even if there are no configuration files on the provisioning server.

For more information, refer to Sample Template Files on page A-6.

Override Files

This file contains all changes that are made by a user through the their phone (for example, time/date formats, ring types, and backlight intensity). The file allows the phone to keep user preferences through reboots and upgrades (providing that the system permits the override file to be written to the provisioning server).

As of Polycom UC Software 3.3.0, separate overrides files are kept for local and web configuration changes.

• There is an option to clear the local override file available to the system administrator—press the Menu key, and then select Settings > Advanced > Admin Settings > Reset to Defaults > Reset Local Configuration. You will be prompted to enter the administrative password.

• There is an option to clear the web override file available to the system administrator—press the Menu key, and then select Settings > Advanced > Admin Settings > Reset to Defaults > Reset Web Configuration. You will be prompted to enter the administrative password.

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Central Provisioning

The phones can be centrally provisioned from a provisioning server through a system of global and per-phone configuration files. The provisioning server also facilitates automated application upgrades, logging, and a measure of fault tolerance. Multiple redundant provisioning 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. 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.

Note

Verify the order of the configuration files. Parameters in the configuration file loaded first will overwrite those in later configuration files.

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

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Resource Files

Overview

Manual Configuration

When the manual configuration method is employed (using the phone’s user interface and/or the web interface), 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 provisioning 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

• Ring tones

• Synthesized tones

• Contact directories

Note

If you need to remove the resource files from a phone at some later date—for example, you are giving the phone to a new user—instructions on how to put the phone into the factory default state can be found in “Quick Tip 18298: Resetting and Rebooting Polycom Phones“ at

http://www.polycom.com/support/voice/soundpoint_ip/VoIP_Technical_Bulletins_pu b.html .

Available Features

This section provides information about the features available on the Polycom phones running Polycom UC Software:

• Basic User Features

— Automatic Off-Hook Call Placement—Supports an optional

automatic off-hook call placement feature for each registration.

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— 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.

— 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 signaling, 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.

— Message Waiting Indication—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.

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Overview

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

stop all incoming call alerting.

— Graphic Display Backgrounds—A picture or design displayed on the

background of the graphic display.

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

standard with a handset and a dedicated headset connection (headset not supplied). All SoundPoint IP, SoundStation IP, and VVX phones have full-duplex speakerphones.

— Idle Display Image Display—All phones 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. The advanced aspects of conferencing are part of the Productivity Suite.

— Local Contact Directory—The phone maintains a local contact

directory that can be downloaded from the provisioning server and edited locally. Any edits to the Contact Directory made on the phone are saved to the provisioning server as a backup.

— 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.

— 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.

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• Advanced Features

— Access URL in SIP Message—Ability for the SoundPoint IP phones to

be able to receive a URL inside a SIP message (for example, as a SIP header extension in a SIP INVITE) and subsequently access that given URL in the Microbrowser.

— SIP-B 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.

— Browser and Microbrowser—The SoundPoint IP 450, 550, 560, 600,

601, 650, and 670 desktop phones, the SoundStation IP 5000, 6000, and 7000 conference phones, and the VVX 1500 phones (pre-SIP 3.2.2) support an XHTML microbrowser. The VVX 1500 phones running SIP

3.2.2 or later support a Webkit browser.

— 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. This feature may require call server support.

— Capturing Phone’s Current Screen—Allows the phone’s current

display to be displayed in a web browser.

— Configurable Feature Keys—Certain key functions can be changed

from the factory defaults.

— Configurable Soft Keys—Allows users to create their own soft keys

and have them displayed with or without the standard SoundPoint IP and SoundStation IP soft keys.

— Corporate Directory—The phone can be configured to access your

corporate directory if it has a standard LDAP interface. This feature is part of the Productivity Suite. Active Directory, OpenLDAP, Microsoft ADAM, and SunLDAP are currently supported.

— Customizable Fonts—The phone’s user interface can be customized

by changing the fonts used on the display and the LED indicator patterns.

— Display of Warnings from SIP Headers—Displays a “pop-up” to user

that is found in the Warning Field from a SIP header.

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

— Enhanced Busy Lamp Field—Allows an attendant to see a remote line

that is Ringing and answer a remote ringing call using a single key-press. Also allows the attendant to view the caller-id of remote active and ringing calls. This feature may require call server support.

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— Enhanced Feature Keys (EFKs)—Allows users to redefine soft keys to

suit their needs. In SIP 3.0, this feature required a license key.

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Overview

— Feature Synchronized Automatic Call Distribution—Supports ACD

agent available and unavailable and allows ACD sign in and sign out. Requires call server support.

— Instant Messaging—Supports sending and receiving instant text

messages.

— LLDP and Supported TLVs—Support for Link Layer Discovery

Protocol (LLDP) and media extensions (LLDP-MED) such as VLAN configuration. For provisioning information, refer to Ethernet Menu on page 3-12.

— Microsoft Live Communications Server 2005

Integration—SoundPoint IP and SoundStation IP phones can be 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 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.

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

allocated to a single registration.

— Multiple Registrations—SoundPoint IP desktop phones and VVX

1500 phones support multiple registrations per phone. However, SoundStation IP conference phones support a single registration.

— 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.

— Quick Setup of Polycom Phones—Simplifies the process of entering

provisioning server parameters.

— 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.

— Recording and Playback of Audio Calls — Recording and playback

allows the user to record any active conversation using the phone on a USB device. The files are date and time stamped for easy archiving and can be played back on the phone or on any computer with a media playback program that supports the .wav format. This feature is part of the Productivity Suite.

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— Server Redundancy—Server redundancy is often required in VoIP

deployments to ensure continuity of phone service for events where the call server needs to be taken offline for maintenance, the server fails, or the connection from the phone to the server fails.

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

logically related to each other. Requires call server support.

— Static DNS Cache—Set up a static DNS cache and provide for negative

caching.

— 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 a wide range of industry 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.

— Dynamic Noise Reduction— Provides maximum microphone

sensitivity, while automatically reducing background noise on SoundStation IP 7000 conference phones.

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

with an 802.1Q VLAN header.

— 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.

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— Low-Delay Audio Packet Transmission—Designed to minimize

latency for audio packet transmission.

— Treble/Bass Controls—Equalizes the tone of the high and low

frequency sound from the speakers on SoundStation IP 7000 conference phones.

— 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.

— Voice Quality Monitoring—Generates various quality metrics

including MOS and R-factor for listening and conversational quality. This feature is part of the Productivity Suite.

• Video Features

— Video Codecs—Support the standard video codecs on the VVX 1500

phones only.

— H.323 Protocol—Support for the H.323 protocol for the VVX 1500

phones only.

• Video Integration Features

Overview

— For more information on how to use the SoundStation IP 7000 with

Polycom HDX systems, refer to the “SoundStation IP 7000 & Polycom HDX Systems” support page at

http://www.polycom.com/support/voice/soundstation_ip_series/ soundstation_ip7000_hdx_series.html .

• 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 provisioning server and it can encrypt files before uploading them to the provisioning server.

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

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

— Digital Certificates— Support for digital certificates and associated

private keys on certain models of Polycom phones.

— Incoming Signaling Validation—Levels of security are provided for

validating incoming network signaling.

— Mutual TLS Authentication—Support for phone authentication of the

server and server authentication of the phone.

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— Secure Real-Time Transport Protocol—Encrypting audio streams to

avoid interception and eavesdropping.

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

4-1.

New Features in Polycom UC Software 3.3.0

Note

Certain phone models (referred to as ‘legacy’ phones) are not supported in the Polycom UC Software 3.3.0 release.

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 2.1.4 .

Any new features introduced after SIP 2.1.4 are not supported. Refer to the SIP 2.1 Administrator Guide, which is available at

http://www.polycom.com/global/documents/support/setup_maintenance/products/v oice/sip_2.1_addendum_to_sip_2.0_administrator%27s_guide.pdf .

The SoundPoint IP 301, 501, 600, and 601 and the SoundStation IP 4000 phones will be supported on the latest maintenance patch release of the SIP 3.1 software stream—currently SIP 3.1.6 . Any new features introduced after 3.1.6 are not supported. Configuration parameters related to these phones will be removed from the sip.cfg and phone1.cfg files in the next major release. To administer these

phones, refer to the SIP 3.1 Administrator’s Guide, which is available at

http://www.polycom.com/voicedocumentation/ .

The SoundPoint IP 430 will be supported on the latest maintenance part release of the SIP 3.2 software stream—currently 3.2.3 . Any new features introduced after

3.2.3 are not supported. Configuration parameters related to these phones have

been removed. To administer these phones, refer to the SIP 3.2 Administrator’s Guide, which is available at http://www.polycom.com/voicedocumentation/ .

The following new features were introduced in SIP 3.2.3:

• DHCP Menu—This feature has been enhanced as follows:

— Support for an alternate format for DHCP option 60 as well as the

introduction of DHCP Vendor Identifying Options 43 and 125.

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The following existing features were changed in SIP 3.2.3:

• Audible Ringer Location—Allows the user to change where incoming call

ringing plays out.

• Server Redundancy—Polycom phones have a “failover” feature that

enables them to re-register before diverting SIP signaling to an alternate server.

• A new loud ringer .wav file—Warble.wav—was introduced.

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Overview

The following new features were introduced in Polycom UC Software 3.3.0:

• Support for EAPOL Logoff Message—The PC Ethernet port on the

SoundPoint IP 33x, 450, 550, 560, 601, 650, and 670, and Polycom VVX 1500 phone can be used to connect a computer to the network with the phone acting as a pass-through switch.

• Locking the Phone—Protects unattended phones, but allows authorized

and emergency calls to be placed.

• Configurable TLS Cipher Suites—Allow administrators to configure

which cipher suites may be used for TLS connections, including support for the ‘null’ cipher which is useful for troubleshooting purposes.

• Calling Party Identification—These features have been enhanced as

follows:

— During the ‘ringing’ stage of an incoming call on the SoundPoint IP

331 and 335, the caller ID will automatically scroll. Auto-scrolling stops— the left and right arrow keys can be used to scroll—once the call is connected.

The following existing features were changed in Polycom UC Software 3.3.0:

Note

• Configuration Parameters—Significant changes to the configuration

system in this release. For a brief introduction, refer to “Technical Bulletin 56449: Polycom SoundPoint IP/SoundStation IP/VVX Software Changes”. For more detailed information, refer to “Technical Bulletin 60519: Simplified Configuration Enhancements in Polycom UC Software

3.3.0”. Both technical bulletins can be found at

http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoI P_Technical_Bulletins_pub.html

• Enhanced Feature Keys—Supported on SoundStation IP 5000, 6000, and

7000 conference phones.

• Local / Centralized Conferencing—The behavior of Polycom phones

when the conference host exits a conference is now configurable:

— The remaining parties are left connected and can continue to talk. This

is the previous behavior.

— All parties are disconnected from the conference.

Documentation of the newly released SoundStation IP 5000 conference phone has also been added.

When SoundPoint IP 32x/33x is used in this guide, it includes the SoundPoint IP 320, 321, 330, 331, and 335 phones.

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Setting up Your System

Your Polycom® 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 Polycom phones. This chapter contains information on:

• Setting Up the Network

• Setting Up the Provisioning Server

• Deploying Phones From the Provisioning Server

• Upgrading Polycom UC Software

Because of the large number of optional installations and configurations that are available, this chapter focuses on one particular way that the Polycom® UC Software 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 phones, the use of a provisioning 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 provisioning server when it boots up, it will operate with internally saved parameters. This is useful for occasions when the provisioning server is not available, but is not intended to be used for long-term operation of the phones.

However, if you want to register a single Polycom phone, refer to “Quick Tip 44011: Register Standalone Polycom Phones“ at

http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_Technical_Bulle tins_pub.html .

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This chapter also contains information on:

• Provisioning SoundStation IP 7000 Phones Using C-Link

• Provisioning VVX 1500 Phones Using a Polycom CMA System

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.

Polycom UC Software uses the network to query the provisioning 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 <MACaddress>.cfg file, and UC Software, 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 3-6.

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.

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Setting up Your System

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 DHCP Option DHCP

 priority when more than one source exists 

12 3 4

IP address 1•-- • subnet mask 1•-- • IP gateway 3•-- •

Refer to DHCP

boot server address

SIP server address SNTP server

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

address

Menu on page 3-8

151

Note: This value is configurable.

42 then 4 • - • •

6•-- •

•• - •

•- - •

DHCP

Configuration File (application only)

Local FLASH

alternate DNS server IP address

DNS domain 15 • - - •

VLAN ID

6•-- •

Refer to DHCP

Menu on page 3-8

Warning: Link Layer Discovery Protocol (LLDP) overrides Cisco Discovery Protocol (CDP). CDP overrides Local FLASH which overrides DHCP VLAN Discovery.

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.

Note

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

The CDP Compatibility value can be obtained from a connected Ethernet switch if the switch supports CDP.

in <sntp/> on page A-113.

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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 provisioning 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.

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 provisioning server depends on several factors including the phone model and whether the BootROM or Polycom UC Software 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 3-11). The boot server address can be an IP address, domain string name, or URL. The boot server address can also be obtained through DHCP. Configuration file names in the <MACaddress>.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.

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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 3-11).

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Protocol used by BootROM

Setting up Your System

Protocol used by Polycom UC Software

Note

IP 32x, 33x, 450,

550, 560, 650, 670, Specified Protocol

FTP FTP FTP FTP

TFTP TFTP TFTP TFTP

HTTP HTTP HTTP HTTP

HTTPS HTTP HTTPS HTTPS

There are two types of FTP methods—active and passive. UC Software is not compatible with active FTP. Secure provisioning was implemented in a previous release.

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.

Both digest and basic authentication are supported when using HTTP/S for Polycom UC Software. Only digest authentication is supported when using HTTP by the BootROM. If the Server Type is configured as HTTPS, the BootROM will contact the same address and apply the same username and password to authentication challenges only the protocol used will be HTTP. No SSL negotiation will take place, so servers that do not allow unsecured HTTP connections will not be able to provision files.

5000, 6000, and

7000

VVX 1500

IP 32x, 33x, 450, 550, 560, 650, 670, 5000, 6000, and 7000, VVX 1500

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 (refer to Trusted Certificate Authority List on page C-1) and custom certificates can be added to the phone (refer to “Technical Bulletin 17877: Using Custom Certificates With Polycom Phones“ at

http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_T echnical_Bulletins_pub.html .

As of SIP 3.2, Mutual Transport Layer Security (TLS) authentication is available. For more information, refer to Mutual TLS Authentication on page

4-97.

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Note

If you want to use digest authentication against the Microsoft Internet Information Services server:

• Use Microsoft Internet Information Server 6.0 or later.

• Digest authentication needs the user name and password to be saved in

reversible encryption.

• The user account on the server must have administrative privileges.

• The wildcard must be set as MIME type; otherwise the phone will not download

*.cfg, *.ld and other required files. This is due to the fact that the Microsoft Internet Information Server cannot recognize these extensions and will return a “File not found” error. To configure wildcard for MIME type, refer to

http://support.microsoft.com/kb/326965 .

For more information, refer to

http://www.microsoft.com/technet/prodtechnol/WindowsServer2003/Library/IIS/809 552a3-3473-48a7-9683-c6df0cdfda21.mspx?mfr=true .

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 Polycom UC Software 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. Polycom recommends that you change the administrative password from the default value.

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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.

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Setting up Your System

Resetting to Factory Defaults

The basic network configuration referred to in the subsequent sections can be reset to factory defaults using a menu selection from the Advanced Settings menu or using a multiple key combination described in Multiple Key

Combinations on page C-9.

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 3-8.

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 3-11. 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.

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.

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Name Possible Values Description Ethernet Refer to Ethernet Menu on page 3-12. EM Power Enabled, Disabled This parameter is relevant if the phone gets Power over

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

Syslog Refer to Syslog Menu on page 3-13.

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 650 and 670 phones.

To switch the text entry mode on the SoundPoint IP 32x/33x, 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:

Name Possible Values Description

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

response received from the DHCP server. The DHCP server should send address information in option 66 that matches one of the formats described for Server Address in the next section, Server Menu.

If the DHCP server sends nothing, the following scenarios are possible:

• If a boot server value is stored in flash memory and the

value is not “0.0.0.0”, then the value stored in flash is used.

• Otherwise the phone sends out a DHCP INFORM query.

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- If 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.

- If no alternate DHCP server responds, the INFORM query process will retry and eventually time out.

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Setting up Your System

Name Possible 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 following scenarios are possible:

• If a boot server value is stored in flash memory and the

value is not “0.0.0.0”, then the value stored in flash is used.

• Otherwise the phone sends out a DHCP INFORM query.

- If 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.

- If no alternate DHCP server responds, the INFORM query process will retry and eventually time out.

2=Static The phone will use the boot server configured through the

Server Menu. For more information, refer to the next 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 following scenarios are

possible:

• If a boot server value is stored in flash memory and the

value is not “0.0.0.0”, then the value stored in flash is used.

• Otherwise the phone sends out a DHCP INFORM query.

- If 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. 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.

- If no alternate DHCP server responds, the INFORM query process will retry and eventually time out.

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 next section, Server

Menu.

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Name 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 the

VLAN ID Option 128 through 254

(Cannot be the same as Boot Server Option)

(default is 129)

Option 60 Format 0=RFC 3925

Binary

1=ASCII String Vendor identifying information in ASCII.

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-21.

Vendor identifying information in the format defined in RFC 3925, which can be found at http://tools.ietf.org/html/rfc3925 .

For more information, refer to “Technical Bulletin 54041: Using DHCP Vendor Identifying Options With Polycom Phones” at

http://www.polycom.com/support/voice/soundpoint_ip/VoIP_Te chnical_Bulletins_pub.html .

Note: DHCP option 125 containing the RFC 3295 formatted data will be sent whenever option 60 is sent.

Note: DHCP option 43 data is ignored.

Note: DHCP option 125 containing the RFC 3295 formatted

data will be sent whenever option 60 is sent. Note: DHCP option 43 data is interpreted as encapsulated

DHCP options and these will take precedence over options received outside of option 43.

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Note

If multiple alternate DHCP servers respond:

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

• If configured for

Custom+Option66

, the phone will select the first response that

contains a valid "custom" option value.

• If none of the responses contain a "custom" option value, the phone will select

the first response that contains a valid “option66” value.

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Server Menu

The following server configuration parameters can be modified on the Server menu:

Name Possible Values Description

Setting up Your System

Server Type 0=FTP, 1=TFTP, 2=HTTP,

3=HTTPS, 4=FTPS, 5=Invalid

Server Address dotted-decimal IP address

OR domain name string OR URL

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

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

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

The protocol that the phone will use to obtain configuration and phone application files from the provisioning 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. Note: Only implicit FTPS is supported.

The provisioning 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 provisioning 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.

(if required) for the selected Server Type. Note: If the Server Address is a URL with a user name,

this will be ignored.

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

Retry Wait 0 to 300

Default 1

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

The minimum amount of time that must elapse before retrying a file transfer, in seconds. The time is measured from the start of a transfer attempt which is defined as the set of upload/download transactions made with the IP addresses that map to a given provisioning 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

Provisioning Server on page 3-17.

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Name Possible Values Description

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.

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

LLDP Enabled, Disabled If enabled, the phone will use the LLDP protocol to

communicate with the network switch for certain network parameters. Most often this will be used to set the VLAN that the phone should use for voice traffic. It also reports power management to the switch. The default value is Enabled.

If the switch does not support it, VLAN Discovery is used. Refer to DHCP Menu on page 3-8.

There are four ways to get VLAN on the phone and they can all be turned on, but the VLAN used is chosen by priority of each method. The priority is: 1. LLDP; 2. CDP;

3. DVD (VLAN Via DHCP); 4. Static (VLAN ID entered in config menu).

For more information, refer to LLDP and Supported TLVs on page C-28.

CDP Compatibility Enabled, Disabled If enabled, the phone will use CDP compatible signaling to

communicate with the network switch for certain network parameters. Most often this will be used to set the VLAN that the phone should use for Voice Traffic, and for the phone to communicate its PoE power requirements 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 Disabled.

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Setting up Your System

Name Possible Values Description

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 5 = 1000FD

PC Port Mode 0 = Auto

1 = 10HD 2 = 10FD 3 = 100HD 4 = 100FD 5 = 1000FD

-1 = Disabled

1000BT LAN Clock 0=Auto

1=Slave 2=Master

1000BT PC Clock 0=Auto

1=Slave 2=Master

The network speed over the Ethernet. The default value is Auto. HD means half duplex and FD means full duplex.

Note: Polycom recommends that you do not change this setting.

The network speed over the Ethernet. The default value is Auto. HD means half duplex and FD means full duplex.

Note: Polycom recommends that you do not change this setting unless you want to disable the PC port.

The mode of the LAN clock. The default value is Slave (this device receives its clock

timing from a master device).

Note: Polycom recommends that you do not change this setting unless you are having Ethernet connectivity issues. This setting was chosen to give the best results from an EMI perspective.

The mode of the PC clock. The default value is Auto.

Note: Polycom recommends that you do not change this setting unless you are having Ethernet connectivity issues. This setting was chosen to give the best results from an EMI perspective.

Note

The LAN Port Mode applies to all phones supported by SIP 3.0 . The PC Port Mode parameters are only available on phones with a second Ethernet port.

Only the SoundPoint IP 560 and 670 and VVX 1500 phones supports the LAN Port Mode and PC Port Mode setting of 1000FD.

The 1000BT LAN Clock and 1000BT PC Clock parameters are only available on SoundPoint IP 560 and 670 phones.

Syslog Menu

Syslog is a standard for forwarding log messages in an IP network. The term “syslog” is often used for both the actual syslog protocol, as well as the application or library sending syslog messages.

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The syslog protocol is a very simplistic protocol: the syslog sender sends a small textual message (less than 1024 bytes) to the syslog receiver. The receiver is commonly called “syslogd”, “syslog daemon” or “syslog server”. Syslog messages can be sent through UDP, TCP, or TLS. The data is sent in cleartext.

Syslog is supported by a wide variety of devices and receivers. Because of this, syslog can be used to integrate log data from many different types of systems into a central repository.

The 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 0 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 <log/> on page D-4.

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 Provisioning Server

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The provisioning server can be on the local LAN or anywhere on the Internet.

Multiple provisioning servers can be configured by having the provisioning server DNS name map to multiple IP addresses. The default number of provisioning servers is one and the maximum number is eight. The following protocols are supported for redundant provisioning servers: HTTPS, HTTP, and FTP. For more information on the protocol used on each platform, refer to

Supported Provisioning Protocols on page 3-4.

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Setting up Your System

All of the provisioning 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 3-11 can be used to configure the number of times each server will be tried for a file transfer and also how long to wait between each attempt. The maximum number of servers to be tried is configurable. For more information, contact your Certified Polycom Reseller.

Note

Be aware of how logs, overrides and directories are uploaded to servers that map 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 <MAC-address>.cfg .

To set up the provisioning server:

Use this procedure as a recommendation if this is your first provisioning server setup.

1. Install a provisioning server application or locate suitable existing server(s).

Polycom recommends that you use RFC-compliant servers.

Note

2. Create an account and home directory.

If the provisioning protocol requires an account name and password, the server account name and password must match those configured in the phones. Defaults are: provisioning protocol: FTP, name: PlcmSpIp, password: PlcmSpIp.

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.

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Note

Typically all phones are configured with the same server account, but the server account provides a means of conveniently partitioning the configuration. Give each account an unique home directory on the server and change the configuration on an account-by-account basis.

3. Copy all files from the distribution zip file to the phone home directory.

Maintain the same folder hierarchy.

There are two distribution zip files. The combined image file contains:

—sip.ld

— a number of template files (for example, sip-basic.cfg and

reg-basic.cfg can be found in the Config folder)

— 000000000000.cfg

— 000000000000-directory~.xml

— SoundPointIP-dictionary.xml (one for each supported language)

— SoundPointIPWelcome.wav

The split image file contains individual sip.ld files for each model as well as the template files and dictionary files.

Refer to the latest Release Notes for a detailed description of each file in the distribution and further information on determining which distribution to use.

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Provisioning Server Security Policy

You must decide on a provisioning server security policy.

Polycom recommends allowing file uploads to the provisioning 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. This greatly eases our ability to provide customer support in diagnosing issues that may occur with the phone operation.

For organizational purposes, configuring a separate log file directory, override directory, contact directory, and license directory is recommended, but not required. The different directories can have different access permissions. For example, for LOG, CONTACTS, and OVERRIDES, allow full access (read and write) and for all others, read-only access. For more information on LOG_FILE_DIRECTORY, OVERRIDES, CONTACTS, and LICENSE, 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.

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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-address>.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 Provisioning Server

You can successfully deploy Polycom phones from one or more provisioning servers.

For all Polycom phones, follow the normal provisioning process in the next section, Provisioning Phones. If you have decided to daisy-chain two SoundStation IP 7000 conference phones together, read the information in

Provisioning SoundStation IP 7000 Phones Using C-Link on page 3-24 to

understand the different provisioning options available. If your organization uses the Polycom® Converged Management Application™ (CMA™) system, read the information in Provisioning VVX 1500 Phones Using a Polycom CMA

System on page 3-25 to understand the different provisioning option available

for your organization’s VVX 1500 phones.

Setting up Your System

Provisioning Phones

As of Polycom UC Software 3.3.0, Polycom phones will start up without any configuration files; however, certain parameters will need to be changed for your phones to be usable within your organization (for example, registration address and label, and SIP server address). You can create as many configuration files as you want; you may want to put SIP server parameters in one file and enhanced feature key definitions on another file. If you want, you can put all parameters into one file.

These changes can be made through one of the following methods:

• Using configuration files hosted on a provisioning server

• Using a web browser to access the phone’s web interface

• Using the phone’s local user interface

For large scale deployments, the configuration file method is strongly recommended. For smaller scale deployments, the phone web interface or local interface may be used, but administrators need to be aware that settings made by these methods will take precedence over centralized configuration files.

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For more information on creating configuration files, refer to the “Configuration File Management on Polycom Phones” white paper at

http://www.polycom.com/global/documents/support/technical/products /voice/white_paper_configuration_file_management_on_soundpoint_ip_ph ones.pdf .

For more information on phone configuration and provisioning, refer to the appropriate Technical Bulletins and Quick Tips at

http://www.polycom.com/support/voice/soundpoint_ip/VoIP_Technical _Bulletins_pub.html .

For more information on encrypting configuration files, refer to Encrypting

Configuration Files on page C-4.

To deploy phones from the provisioning server:

1. Create per-phone configuration file(s) by performing the following steps:

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.

For more information on template files included with Polycom UC software 3.3.0, refer to Sample Template Files on page A-6.

Note

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

Do not use [MACaddress]-phone.cfg as the per-phone filename. This filename is used by the phone itself to store user preferences (overrides).

For example, add phone registration parameters.

2. Create per-site configuration file(s) by performing the following steps:

a Create per-site site[location].cfg file.

For more information on template files included with Polycom UC software 3.3.0, refer to Sample Template Files on page A-6.

For example, add the SIP server or feature parameters.

Most of the default settings are typically adequate, however, if SNTP settings are not available through DHCP, the SNTP GMT offset and (possibly) the SNTP server address will need to be edited for the correct local conditions. Changing the default daylight savings parameters will likely be necessary outside of North American locations. (Optional) Disable the local web (HTTP) server or change its signaling port if local security policy dictates (refer to <httpd/> on page A-63). Change the default location settings for user interface language and time and date format (refer to <lcl/> on page A-65).

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Setting up Your System

3. Create a master configuration file by performing the following steps:

a Create per-phone or per-platform <MACaddress>.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 <MACaddress>.cfg files so

that it references the appropriate configuration file(s).

For example, add a reference to phone[MACaddress].cfg and sip650.cfg.

c Edit the LOG_FILE_DIRECTORY attribute of the <MACaddress>.cfg

files so that it points to the log file directory.

d Edit the CONTACT_DIRECTORY attribute of the

<MACaddress>.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 UC Software modify the APPLICATION APP_FILE_PATH attribute of the <MACaddress>.cfg files so that it references the appropriate sip.ld files.

Note

For example, the reference to sip.ld is changed to 2345-11670-001.sip.ld to boot the SoundPoint IP 670 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-22.

5. Ensure that the configuration process completed correctly.

For example, on the phone, press the Menu key, and then select Status >

Platform > Application to see the UC Software version and Status > Platform > Configuration to see the configuration files downloaded to the

phone.

Monitor the provisioning server event log and the uploaded event log files (if permitted). All configuration files used by the provisioning server are logged.

You can now instruct your users to start making calls.

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Upgrading Polycom UC Software

You can upgrade the software that is running on the Polycom phones in your organization. The exact steps that you perform are dependent on the version of Polycom UC Software that is currently running on the phones and the version that you 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 Current SoundPoint IP, SoundStation IP, and VVX Phones. If you

provisioned your VVX phones using CMA, refer to Upgrading Polycom UC

Software Using Polycom CMA on page 3-27.

However, if your organization has a mixture of legacy phones—for example, SoundPoint IP 300, 301, 430, 500, 501, 600, 601 and/or SoundStation IP 4000 phones—deployed along with other models, you will need to change the phone configuration files to continue to support the SoundPoint IP 300, 301, 430, 500, 501, 600, and 601 and SoundStation IP 4000 phones when software releases UC Software 3.3.0 or later are deployed. These models were discontinued as follows:

Warning

• The SoundPoint IP 300 and 500 phones as of May 2006

• The SoundPoint IP 301, 600, and 601 phones as March 2008

• The SoundPoint IP 501 phone as of August 2009

• The SoundStation IP 4000 phone as of May 2009

• The SoundPoint IP 430 phone as of April 2010

In all cases, refer to Supporting Legacy SoundPoint IP and SoundStation IP

Phones on page 3-22.

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 2.1.4. 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.

The SoundPoint IP 301, 501, 600, and 601 and the SoundStation IP 4000 phones will be supported on the latest maintenance patch release of the SIP 3.1 software stream—currently SIP 3.1.3 . 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 or later for these changes to be effective.

The SoundPoint IP 430 phone will be supported on the latest maintenance patch release of the SIP 3.2 software stream—currently SIP 3.2.3. Any critical issues that affect SoundPoint IP 430 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 4.2.2 for these changes to be effective.

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Setting up Your System

Supporting Current SoundPoint IP, SoundStation IP, and VVX Phones

Warning

If you need to upgrade any VVX 1500 phones running SIP 3.1.3 or earlier to SIP

3.2.2 or later, you must perform additional steps before rebooting the phone to download the software. Refer to “Technical Bulletin 53522: Upgrading the VVX 1500 Phone to SIP 3.2.2” at

http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_Technical_Bulle tins_pub.html .

To update phones to Polycom UC Software 3.3.0:

1. Back up old application and configuration files.

2. Create new configuration files.

Differences between old and new versions of configuration files are explained in the Release Notes that accompany the software. All changes are mandatory for upgrading to UC Software 3.3.0.

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 Polycom

Phones” white paper at

http://www.polycom.com/global/documents/support/technical/products/voice/white_ paper_configuration_file_management_on_soundpoint_ip_phones.pdf .

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

4. Reboot the phones using automatic methods such as polling or check-sync.

Using the reboot multiple key combination should be done as a backup option only. For more information, refer to Multiple Key Combinations on page C-9.

Since the APPLICATION APP_FILE_PATH attribute of the <MACaddress>.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 2345-11670-001.sip.ld to boot the SoundPoint IP 670 image.

The phones can be rebooted remotely through the SIP signaling protocol. Refer to <specialEvent/> on page A-156.

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The phones can be configured to periodically poll the provisioning 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 <prov/> on page A-78.

Supporting Legacy SoundPoint IP and SoundStation IP Phones

With enhancements available since BootROM 4.0.0 and SIP 2.1.2, you can modify the 000000000000.cfg or <MACaddress>.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.

Polycom UC Software 3.3.0 or later software distributions contain only the new distribution files for the new release. You must rename the sip.ld, sip.cfg, and phone1.cfg from a previous 2.1.x distribution that is compatible with SoundPoint IP 300 and 500 phones or a previous 3.1.y distribution that is compatible with SoundPoint IP 301, 501, 600, and 601 SoundStation IP 4000 phones or a previous 3.2.z distribution that is compatible with SoundPoint IP 430 phones.

The following procedure must be used for upgrading to UC Software 3.3.0 or later for installations that have SoundPoint IP 300, 301, 430, 500, 501, 600, 601 and SoundStation IP 4000 phones deployed. It is also recommended that this same approach be followed even if these phones are not part of the deployment as it will simplify management of phone systems with future software releases.

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To upgrade phones to Polycom UC Software 3.3.0:

1. Do one of the following steps:

a Place all bootrom.ld files corresponding to BootROM release zip file

onto the provisioning server.

b Ensure that all phones are running BootROM 4.0.0 or later code.

2. Copy sip.ld (or the appropriate individual sip.ld from the split image

file) from the UC Software 3.3.0 or later release distribution onto the provisioning server.

These are the relevant files for all phones except the SoundPoint IP 300, 301, 430, 500, 501, 600, 601 and SoundStation IP 4000 phones.

3. Rename sip.ld, sip.cfg, and phone1.cfg from the previous distribution to sip_21x.ld, sip_21x.cfg, and phone1_21x.cfg respectively on the

provisioning server.

These are the relevant files for supporting the SoundPoint IP 300 and 500 phones.

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4. Rename sip.ld, sip.cfg, and phone1.cfg from the previous distribution to sip_31y.ld, sip_31y.cfg, and phone1_31y.cfg respectively on the

provisioning server.

These are the relevant files for supporting the SoundPoint IP 301, 501, 600, 601 and SoundStation IP 4000 phones.

5. Rename sip.ld, sip.cfg, and phone1.cfg from the previous distribution to sip_323.ld, sip_323.cfg, and phone1_323.cfg respectively on the

provisioning server.

These are the relevant files for supporting the SoundPoint IP 430 phones.

6. 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_214.ld" APP_FILE_PATH_SPIP300="sip_214.ld" APP_FILE_PATH_SPIP601="sip_313.ld" APP_FILE_PATH_SPIP600="sip_313.ld" APP_FILE_PATH_SPIP501="sip_313.ld" APP_FILE_PATH_SPIP301="sip_313.ld" APP_FILE_PATH_SSIP4000="sip_313.ld" APP_FILE_PATH_SPIP430="sip_323.ld" CONFIG_FILES="[PHONE_MAC_ADDRESS]-user.cfg, phone1.cfg, sip.cfg" CONFIG_FILES_SPIP500="[PHONE_MAC_ADDRESS]-user.cfg,

phone1_214.cfg, sip_214.cfg"

CONFIG_FILES_SPIP300="[PHONE_MAC_ADDRESS]-user.cfg,

phone1_214.cfg, sip_214.cfg"

CONFIG_FILES_SPIP601="[PHONE_MAC_ADDRESS]-user.cfg,

phone1_313.cfg, sip_313.cfg"

CONFIG_FILES_SPIP600="[PHONE_MAC_ADDRESS]-user.cfg,

phone1_313.cfg, sip_313.cfg"

CONFIG_FILES_SPIP501="[PHONE_MAC_ADDRESS]-user.cfg,

phone1_313.cfg, sip_313.cfg"

CONFIG_FILES_SPIP301="[PHONE_MAC_ADDRESS]-user.cfg,

phone1_313.cfg, sip_313.cfg"

CONFIG_FILES_SSIP4000="[PHONE_MAC_ADDRESS]-user.cfg,

phone1_313.cfg, sip_313.cfg"

CONFIG_FILES_SPIP430="[PHONE_MAC_ADDRESS]-user.cfg,

phone1_323.cfg, sip_323.cfg"

MISC_FILES="" LOG_FILE_DIRECTORY="" OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY="" />

7. Remove any <MACaddress>.cfg files that may have been used with earlier releases from the provisioning server.

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Note

This approach takes advantage of an enhancement that was added in BootROM 3.2.1/SIP 2.0.1 that allows for the substitution of the phone specific [MACADDRESS] inside configuration files. This avoids the need to create unique <MACaddress>.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 <MACaddress>.cfg files for SoundPoint IP 300, 301, 430, 500, 501, 600, and 601 and SoundStation IP 4000 phones or all of the <MACaddress>.cfg files if it is not explicitly known which phones are SoundPoint IP 300, 301, 430, 500, 501, 600, and 601 and SoundStation IP 4000 phones.

For more information, refer to “Technical Bulletin 35311: Supporting Legacy Polycom Phones with SIP 2.2.0, SIP 3.2.0, or Polycom UC Software 3.3.0 and Later Releases“ at

http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_T echnical_Bulletins_pub.html .

Provisioning SoundStation IP 7000 Phones Using C-Link

Normally the SoundStation IP 7000 conference phone is provisioned over the Ethernet by the provisioning server. However, when two SoundStation IP 7000 phones are daisy-chained together, the one that is not directly connected to the Ethernet can still be provisioned (known as the secondary).

Power Adapter

Multi-Interface Module

12-foot Ethernet Cable

Interconnect Cable

25-foot Network Cable

The provisioning over C-Link feature is automatically enabled when a SoundStation IP 7000 phone is not connected to the Ethernet. Both SoundStation IP 7000 phones must be running the same version of Polycom UC Software.

The steps for provisioning the secondary SoundStation IP 7000 phone are the same as for the primary SoundStation IP 7000 phone. You can reboot the primary without rebooting the secondary. However, the primary and

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secondary should be rebooted together for the primary/secondary relationship to be recognized. If you power up both SoundStation IP 7000 phones, the primary will power up first.

Currently, provisioning over C-Link is supported for the following configurations of SoundStation IP 7000 conference phones:

• Two SoundStation IP 7000 conference phone daisy-chained together

• Two SoundStation IP 7000 conference phone daisy-chained together with

one external microphone, specifically designed for the SoundStation IP 7000 conference phone

The provisioning server (or proxy) for the secondary is determined by the following criteria:

• The primary phone must be powered up using Multi-Interface Module. PoE will not provide enough power for both phones.

• If the secondary is configured for DHCP, use the primary’s provisioning server if the primary is configured for DHCP.

• If the secondary is not configured for DHCP, use the secondary’s static provisioning server if it exists.

• If the secondary’s static provisioning server does not exists, use the primary’s provisioning server (ignoring the source).

For more information on daisy-chaining and setting up the SoundStation IP 7000 conference phone, refer to the Setup Guide for the Polycom SoundStation IP 7000 Phone, which is available at

http://www.polycom.com/voicedocumentation/.

Provisioning VVX 1500 Phones Using a Polycom CMA System

Note

This functionality will be available in a future patch release.

You can provision your organization’s VVX 1500 phones and update the software using a Polycom CMA system. Refer to the latest Release Notes for Polycom UC Software and Polycom CMA for specific compatibility requirements and recommendations.

You can also provision your organization’s VVX 1500 phones in a hybrid model, using both Polycom CMA and a provisioning server. In such a situation, Polycom CMA has a higher priority. When the phone reboots, it will check the Polycom CMA system first for new software, and then checks the provisioning server for configuration files and directories to upload if directed to do so (by setting the CMA mode to Disable, refer to Disabling Provisioning

by Polycom CMA System on page 3-27).

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In dynamic management mode, the Polycom CMA system can do the following:

• Configure VVX 1500 phones using an automatic provisioning service

• Register VVX 1500 phones with a standard-based presence service, so that

presence states are shared with Polycom CMA contacts

• Provide VVX 1500 phones with automatic software updates

This section contains information on:

• Provisioning Using Polycom CMA

• Upgrading Polycom UC Software Using Polycom CMA

• Monitoring by Polycom CMA

Provisioning Using Polycom CMA

Note

To be provisioned by the Polycom CMA system, the VVX phones must be running at least Polycom UC Software 3.3.0 .

Polycom CMA requires that the management application be installed on the same network to which your VVX 1500 phones are connected.

To configure the provisioning service settings on VVX 1500 phones:

1. Press the Menu key, and then select Settings > Advanced >

Admininstration Settings > Network Configuration > CMA Menu.

You must enter the administrator password to access the network configuration. The factory default password is 456.

2. Enter the following values:

— CMA Mode: Select Static or Auto.

— Server Address: Enter the address of the Polycom CMA system

running the provisioning service. The address can be an IP address or a fully qualified domain name. For example, 123.45.67.890 .

3. Scroll to Login Credentials and tap the Select soft key. Enter the following values:

—CMA Domain: Enter the domain for registering to the provisioning

service. For example, NorthAmerica .

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Note

If you are not using a Single Sign On login with Active Directory on the Polycom CMA system, the domain will be local using the local accounts created on the Polycom CMA server.

—CMA User: Enter the username for registering to the provisioning

service. For example, bsmith .

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— CMA Password: Enter the password that registers the VX 1500 phone

to the provisioning service (associated with the CMA user account). For example, 123456 .

4. Tap the Back soft key three times.

5. Select Save Config.

The VVX 1500 phone reboots.

Note

Only one phone line associated with a Polycom CMA system can be provisioned on a VVX 1500 phone, but the line key associated with that line is configurable. For more information on configuration file settings, refer to <prov/> on page A-78.

The user can now search for CMA users and groups in the CMA directory, place calls to those contacts, and view their presence status. For more information, refer to the User Guide for the Polycom VVX 1500 Phone at

http://www.polycom.com/support/vvx1500 .

For more information about provisioning by a Polycom CMA system, refer to the Polycom CMA System Deploying Visual Communications Administration Guide and Polycom CMA System Operations Guide, which are available at

http://www.polycom/support/cma_4000_5000 .

Disabling Provisioning by Polycom CMA System

To disable provisioning of the VVX 1500 phones by the Polycom CMA system:

1. Press the Menu key, and then select Settings > Advanced >

Admininstration Settings > Network Configuration > CMA Menu.

You must enter the administrator password to access the network configuration. The factory default password is 456.

2. Enter the following values:

— CMA Mode: Select Disable.

3. Tap the Back soft key twice.

4. Select Save Config.

The VVX 1500 phone reboots.

Upgrading Polycom UC Software Using Polycom CMA

Software upgrades of the VVX 1500 phones are triggered by the Polycom CMA system as either automatic or scheduled updates.

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Note

Software update timer changes will not take effect until the next interval—after the current interval expires. For example:

• The current software update timer is set to 60 minutes.

• The provisioning by the Polycom CMA system fails.

• The software update timer is reset to five minutes (default).

The five minute timer is not fired off until the last 60 minutes timer expires.

For more information about software updates from the Polycom CMA system, refer to the Polycom CMA System Deploying Visual Communications Administration Guide and Polycom CMA System Operations Guide, which are available at http://www.polycom/support/cma_4000_5000 .

Monitoring by Polycom CMA

The following information is sent by the VVX 1500 phone to the Polycom CMA system :

• Network adapter probe—This is the first message that the VVX 1500 phone sends to the Polycom CMA system. It provides the phone’s IP address.

• Software update check—This message provides the phone model, MAC address, and UC Software version currently running on the phone.

• Software update status—This message provides confirmation of the phone’s software upgrade.

• Provisioning profile—This message requests configuration data for the phone so that the user can access the CMA directory, add CMA contacts to their Buddy list, and places audio and video calls to those contacts.

• Provisioning status—This message provides confirmation of the receipt of the configuration data from the Polycom CMA system.

• Call statistics—These messages are sent for all calls placed or answered by the phone’s user.

• Call end—This message is sent after all calls have ended.

• Heartbeat data—This message is sent to the Polycom CMA system

periodically. How often the message is sent is configured by the administrator of the Polycom CMA system.

• Events— This message provides information like gatekeeper registration events, presence registration events, and LDAP events to the Polycom CMA system.

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Configuring Your System

After you set up your Polycom® phones on the network, you can allow users to place and answer calls using the default configuration, however, you may 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 Video Features

• Setting Up Security Features

This chapter also provides instructions on:

• Configuring Polycom Phones Locally

To troubleshoot any problems with your Polycom phones on the network, refer to Troubleshooting Your Polycom 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

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• Missed Call Notification

• Connected Party Identification

• 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 Image Display

• Ethernet Switch

• Graphic Display Backgrounds

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

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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 (provisioning

server)

Call Timer

Call Waiting

Note

Configuration template:

features.cfg

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 32x/33x and SoundStation IP 7000.

and received calls appear as .

Configuration changes can be performed centrally at the provisioning server:

Enable or disable all call lists or individual call lists.

• Refer to <feature/> on page A-58.

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.

Central (provisioning

server)

When an incoming call arrives while the user is active on another call, the incoming call is presented to the user visually on the LCD display. A configurable sound effect such as the familiar call-waiting beep will be mixed with the active call audio as well.

Configuration changes can performed centrally at the provisioning server:

Configuration template:

sip-interop.cfg

Configuration template:

reg-advanced.cfg

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

• Refer to <callWaiting/> on page A-30.

Disable call waiting.

• Refer to <reg/> on page A-82.

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For related configuration changes, refer to Customizable Audio Sound Effects on page 4-74.

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.

The identity displayed is based on the number of the placed call and information obtained from the network signaling.

Note

The phone does not match the number of the placed call to any entries in the Local Contact Directory or Corporate Directory.

There are no related configuration changes.

Calling Party Identification

The phone displays the caller identity, derived from the network signaling, 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 Contact Directory entry may optionally be substituted.

Note

The phone does not match the received number to any entries in the Corporate Directory.

During the ‘ringing’ stage of an incoming call on the SoundPoint IP 331 and 335, the caller ID will automatically scroll as of Polycom® UC Software 3.3.0. Auto-scrolling stops once the call is connected, but the left and right arrow keys can be used to scroll. For example:

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

Central (provisioning

server)

Local Web Server

Configuration templates:

reg-advanced.cfg, site.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 phone can be configured to use a built-in missed call counter or to display information provided by a Session Initiation Protocol (SIP) server.

Note

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

Configuration changes can performed centrally at the provisioning server:

Specify whether or not to use directory name substitution.

• Refer to <up/> on page A-120.

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>-web.cfg on the provisioning server. Changes will permanently override global settings unless deleted through the

Reset Web Configuration menu selection.

Central (provisioning

server)

Configuration template:

features.cfg

Configuration template:

reg-advanced.cfg

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. For example, Bob places a call to Alice, but he ends up connected to Fred. The phone does not match caller IDs against the local contact directory or corporate directory entries.

There are no related configuration changes.

Turn this feature on or off.

• Refer to <feature/> on page A-58.

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

• Refer to <serverMissedCall/> on page A-29.

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Message Waiting Indication

The phone will flash a message-waiting indicator (MWI) LED when instant messages and voice messages are waiting.

Configuration changes can be performed centrally at the provisioning server:

Central (provisioning server)

Configuration template:

sip-interop.cfg

Configuration templates:

reg-advanced.cfg, site.cfg

Specify per-registration whether the MWI LED is enabled or disabled.

• Refer to <mwi/> on page A-72.

Specify whether MWI notification is displayed for registration x (pre-SIP 2.1 behavior is enabled).

• Refer to <up/> on page A-120.

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.

Distinctive Ringing

There are three options for distinctive ringing:

1. The user can select the ring type for each line by pressing the Menu key, and then selecting Settings > Basic > Ring Type. This option has the third (lowest) priority.

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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 is second in priority.

3. The

voIpProt.SIP.alertInfo.x.value

voIpProt.SIP.alertInfo.x.class

and

fields can be used to map calls to specific ring types. This option requires server support and is first (highest) in priority.

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Configuration changes can be performed centrally at the provisioning server or locally:

Central (provisioning

server)

Local Local Phone User

Configuration template:

sip-interop.cfg

Configuration template:

reg-advanced.cfg XML File: <Ethernet

address>-directory. xml

Interface

Distinctive Call Waiting

The

voIpProt.SIP.alertInfo.x.class

call waiting types, currently limited to two styles. This feature requires server support.

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

• Refer to <alertInfo/> on page A-155.

Specify the ring type to be used for each line.

• Refer to <reg/> on page A-82.

This file can be created manually using an XML editor.

• Refer to Local Contact Directory on page 4-9.

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 provisioning server. Changes will permanently override global settings unless deleted through the

Reset Local Configuration menu selection and the <Ethernet address>-phone.cfg is removed from the provisioning server.

voIpProt.SIP.alertInfo.x.value

fields can be used to map calls to distinct

and

Central (provisioning

server)

Do Not Disturb

Note

Configuration changes can be performed centrally at the provisioning server:

Configuration template:

sip-interop.cfg

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 4-22.

A phone with a shared line that has DND enabled will show an incoming call, but the phone will not ring.

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

• Refer to <alertInfo/> on page A-155.

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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 unless configured otherwise.

Server-based DND will behave the same as per-SIP 2.1 per-registration feature with the following exceptions:

• Server based DND cannot be used if the phone is configured as a shared line.

• 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).

Configuration changes can be performed centrally at the provisioning server or locally:

Central (provisioning

server)

Local Local Phone User

Configuration template:

sip-interop.cfg

Configuration template:

reg-advanced.cfg

Interface

Enable or disable server-based DND.

• Refer to <SIP/> on page A-147. Enable or disable local DND behavior when server-based enabled.

• Refer to <SIP/> on page A-147. Specify whether or not DND results in incoming calls being given

busy treatment.

• Refer to <call/> on page A-21. Specify whether DND is treated as a per-registration feature or a

global feature on the phone.

• Refer to <dnd/> on page A-50.

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

• Refer to <reg/>on page A-82.

Enable or disable DND using the Do Not Disturb key on the SoundPoint IP 550, 560, 650, and 670 and the Polycom VVX 1500 or the “Do Not Disturb” option on the Features menu on the SoundPoint IP 32x/33x and 450 and SoundStation IP 5000, 6000 and 7000.

Note: The LED on the Do Not Disturb key on the Polycom VVX 1500 is red when pressed or when server-based DND is enabled.

Handset, Headset, and Speakerphone

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SoundPoint IP phones come standard with a handset and a dedicated connector is provided for a headset (not supplied). All Polycom phones are full-duplex speakerphones. The SoundPoint IP phones provide dedicated keys for convenient selection of either the speakerphone or headset.

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All Polycom desktop phones can be configured to use the electronic hookswitch. For more information, refer to “Technical Bulletin 35150: Using an Electronic Hookswitch with SoundPoint IP and Polycom VVX 1500 Phones“ at

http://www.polycom.com/support/voice/soundpoint_ip/VoIP_Technical _Bulletins_pub.html .

Configuration changes can be performed centrally at the provisioning server or locally:

Central (provisioning

server)

Local Web Server

Configuration templates:

reg-advanced.cfg, site.cfg

(if enabled)

Local Phone User Interface

Enable or disable persistent headset mode.

• Refer to <up/> on page A-120. Enable or disable hands-free speakerphone mode.

• Refer to <up/> on page A-120. Specify whether or not the electronic hookswitch is enabled and what

type of headset is attached.

• Refer to <up/> on page A-120. Switch audio mode from handset to headset or headset to handset.

• Refer to <up/> on page A-120.

Enable or disable persistent headset mode. Navigate to: http://<phoneIPAddress>/coreConf.htm#us Changes are saved to local flash and backed up to <Ethernet

address>-web.cfg on the provisioning server. Changes will permanently override global settings unless deleted through the Reset Web Configuration menu selection.

Enable or disable persistent headset mode through the Settings menu (Settings > Basic > Preferences > Headset > Headset Memory Mode).

Enable or disable hands-free speakerphone mode through the Settings menu (Settings > Advanced > Admin Settings > Phone Settings).

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

Local Contact Directory

The phone maintains a local contact directory. The directory can be downloaded from the provisioning server and edited locally (if configured in that way). 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. The maximum number of entries in the local contact directory is phone-dependent.

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Central (provisioning

server)

Note

If a user makes a change to the local contact directory, there is a five second timeout before it is uploaded to the provisioning server as <mac-address>-directory.xml.

If so configured, the first and last name fields of the local contact directory entries which match incoming calls will be used for caller identification display and in the call lists (instead of the name provided through network signaling).

Configuration changes can be performed centrally at the provisioning server or locally:

Configuration template:

features.cfg

XML file: 000000000000-direct

ory.xml

Specify the maximum number of contacts allowed.

• Refer to <local/> on page A-43. Specify whether or not the local contact directory is read only.

• Refer to <local/> on page A-43. 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 the next section, Local Contact

Directory File Format.

XML file: <Ethernet

address>-directory. xml

Local Local Phone User

Interface

Local Contact Directory File Format

An example of a local contact directory is shown below. The subsequent table provides an explanation of each element. Elements can appear in any order.

<?xml version=”1.0” encoding=”UTF-8” standalone=”yes” ?> <directory>

<item_list>

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This file can be created manually using an XML editor. For information on file format, refer to the next section, Local Contact

Directory File Format.

The user can edit the directory contents if configured in that way. Changes will be stored in the phone’s flash file system and backed up

to the provisioning server copy of <Ethernet address>-directory.xml if this is configured. When the phone boots, the provisioning server copy of the directory, if present, will overwrite the local copy.

<item>

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</item>

... <item>

<ln>Smith</ln> <fn>Bill</fn> <ct>1003</ct> <sd>3</sd> <lb>Dr</lb> <pt>SIP</pt> <rt>3</rt> <dc/> <ad>0</ad> <ar>0</ar> <bw>0</bw> <bb>0</bb>

</item>

</item_list>

</directory>

Configuring Your System

Element Permitted Values Interpretation

fn UTF-8 encoded string

of up to 40 bytes

first name

Note: In some cases, this will be less than 40 characters due to UTF-8’s variable length encoding.

ln UTF-8 encoded string

of up to 40 bytes

last name

Note: In some cases, this will be less than 40 characters due to UTF-8’s variable length encoding.

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. For VVX 1500 phones, the maximum field length is 128 characters; for all other phones, the maximum is 128 characters.

Note: This field cannot be null or duplicated.

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Element Permitted Values Interpretation

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 32x/33x and the SoundStation IP 7000, the maximum speed-dial index is 99.

lb UTF-8 encoded string

of up to 40 bytes

label

Note: In some cases, this will be less than 40 characters due to UTF-8’s variable length encoding.

Note: The label of a contact directory item is by default the label attribute of the item. If the label attribute does not exist or is Null, then the concatenation of first name and last name will be used as label. A space is added between first and last names.

pt “SIP”, “H323”, or

“Unspecified”

protocol The protocol to use when placing a call to this contact.

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

ad 0,1 auto divert

If set to 1, automatically diverts callers that match the directory entry to the address specified in divert contact.

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 buddy 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.

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Local Digit Map

Configuring Your System

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 based on recommendations 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 possible to strip a trailing # from the digits sent or to replace certain matched digits (with the introduction of “R” to the digit map). It is also possible to direct the protocol used to place a call (with the introduction of “S” and “H” to the digit map).

For more detailed information on digit maps, refer to the next section, Digit

Maps.

For more information, refer to “Technical Bulletin 11572: Changes to Local Digit Maps on Polycom Phones“ at

http://www.polycom.com/support/voice/soundpoint_ip/VoIP_Technical _Bulletins_pub.html .

Note

Configuration changes can be performed centrally at the provisioning server or locally:

Central (provisioning

server)

Local Web Server

Configuration template: site.cfg

(if enabled)

Digit maps do not apply to on-hook dialing. The parameter settings described in

<dialplan/> on page A-34 are ignored during on-hook dialing.

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

• Refer to <dialplan/> on page A-34. Specify per-registration impossible match behavior, trailing #

behavior, digit map matching strings, and time out values that override those per-site values.

• Refer to <dialplan/> on page A-34.

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>-web.cfg on the provisioning server. Changes will permanently override global settings unless deleted through the Reset Web Configuration menu selection and the <Ethernet address>-web.cfg is removed from the provisioning server.

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Digit Maps

A digit map is defined either by a “string” or by a list of strings. Each string in the list is an alternative numbering scheme, specified either as a set of digits or timers, or as an expression over which the gateway will attempt to find a shortest possible match.

Digit map extension letter “R” indicates that certain matched strings are replaced. Digit map timer letter “T” indicates a timer expiry. Digit map protocol letters “S” and “H” indicate the protocol to use when placing a call. The following examples shows the semantics of the syntax:

R9RRxxxxxxx

•

—Remove 9 at the beginning of the dialed number

— For example, if a customer dials 914539400, the first 9 is removed

when the call is placed.

•

RR604Rxxxxxxx

—Prepend 604 to all seven digit numbers

— For example, if a customer dials 4539400, 604 is added to the front of

the number, so a call to 6044539400 is placed.

•

R9R604Rxxxxxxx

•

R999R911R

•

xxR601R600Rxx

—Replaces 9 with 604

—Convert 999 to 911

—When applied on 1160122 gives 1160022

• xR60xR600Rxxxxxxx—Any 60x will be replaced with 600 in the middle of the dialed number that matches

— For example, if a customer dials 16092345678, a call is placed to

16002345678.

• 911xxx.T— A period (".") which matches an arbitrary number, including zero, of occurrences of the preceding construct

— For example:

911123 with waiting time to comply with T is a match 9111234 with waiting time to comply with T is a match 91112345 with waiting time to comply with T is a match and the number can grow indefinitely given that pressing the next digit takes less than T.

4 - 14

Note

• 0xxxS|33xxH—All four digit numbers starting with a 0 are placed using the SIP protocol, whereas all four digit numbers starting with 33 are placed using the H.323 protocol.

Only VVX 1500 phones will match the “H”. It is ignored by all other phones and the user will need to press the Send soft key to complete dialing. For example, if the digit map is “33xxH”, the result is as follows:

• If a VVX 1500 user dials “3302” on an H.323 or dual protocol line, the call will be

placed after the user dials the last digit.

• If a SoundPoint IP 650 user dials “3307”, the user must press the Send soft key

to complete dialing.

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Microphone Mute

Configuring Your System

The following guidelines should be noted:

• The letters (x, T, R, S, H) are case sensitive.

• You must use only *, #, +, or 0-9 between second and third R

• If a digit map does not comply, it is not included in the digit plan as a valid

one. That is, no matching is done against it.

• There is no limitation on the number of R triplet sets in a digit map. However, a digit map that contains less than full number of triplet sets (for example, a total of 2Rs or 5Rs) is considered an invalid digit map.

• Using T in the left part of RRR syntax is not recommended. For example, R0TR322R should be avoided.

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.

Using the Configurable Soft Key configuration parameters, an administrator can modify the default soft keys by removing them at different call stages and/or adding specific single or multiple functions. Refer to Enhanced

Feature Keys on page 4-40 and Configurable Soft Keys on page 4-45.

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.

For SoundPoint IP 32x/33x desktop phones and SoundStation IP 6000 and 7000 conference phones, the speed dial index range is 1 to 99. For all other SoundPoint IP and Polycom VVX phones, the range is 1 to 9999.

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 4-60.

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

Central (provisioning

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 Network Time Protocol (SNTP) timeserver. The time and date displayed on the phone will flash continuously to indicate that they are not accurate until a successful SNTP response is received. The time and date display can use one of several different formats and can be turned off. The SoundPoint IP 32x/33x and IP 450 phones have a limited selection of date formats due to a smaller display size.

The

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 6000 are available for selection within the speed dial menu. (Press the Up arrow key from the idle display to jump to the Speed Dial list).

• Refer to Local Contact Directory on page 4-9.

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 provisioning server copy of <Ethernet address>-directory.xml if this is configured. When the phone boots, the provisioning server copy of the directory, if present, will overwrite the local copy.

element in the <Ethernet address>-directory.xml

and 7000) and

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Configuring Your System

Configuration changes can be performed centrally at the provisioning server or locally:

Central (provisioning

server)

Local Web Server

Configuration templates:

reg-advanced.cfg, site.cfg

(if enabled)

Local Phone User Interface

Turn time and date display on or off.

• Refer to <up/> on page A-120. Set the time and date display formats.

• Refer to <datetime/> on page A-68. Set the basic SNTP settings and daylight savings parameters.

• Refer to <sntp/> on page A-113.

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>-web.cfg on the provisioning server. Changes will permanently override global settings unless deleted through the

Reset Web Configuration menu selection and the <Ethernet address>-web.cfg is removed from the provisioning server.

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

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 provisioning server. They will permanently

override global settings unless deleted through the Reset Local Config menu selection.

Idle Display Image Display

All phones can display a customized static image on the idle display in addition to the time and date. For example, a company logo could be displayed (refer to Adding a Customizable Logo on the Idle Display on page

C-6).

As of Polycom UC Software 3.3.0, customized animations are not supported.

Configuration changes can be performed centrally at the provisioning server:

Central (provisioning

server)

Note

Configuration template:

features.cfg

To add an idle display static logo.

• Refer to <bitmap/> on page A-20.

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Ethernet Switch

The SoundPoint IP phones (except the SoundPoint IP 32x) and the VVX 1500 contain two Ethernet ports, labeled LAN and PC, and an embedded Ethernet switch that runs at full line-rate. The SoundStation IP phones contain only one Ethernet port, labeled LAN. 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 disable the PC Ethernet port, refer to Disabling PC Ethernet Port on page C-25.

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<voice/> on page A-134 and <video/> on page A-125.

Graphic Display Backgrounds

You can set up a picture or design to be displayed on the background of the graphic display of all SoundPoint IP 450, 550, 560, 650, and 670 and Polycom VVX 1500 phones.

Note

4 - 18

When installing a background of your choice, care needs to be taken to ensure that the background does not adversely affect the visibility of the text on the phone display. As a general rule, backgrounds should be light in shading for better usability.

For SoundPoint IP 450, 550, 560, 650, and 670 phones:

• There are a number of default backgrounds, both solid color and pictures. Both BMP and JPEG files are supported. You can also select the label color for soft key and line key labels. Users can select which background and label color appears on their phone.

You can modify the supported solid color and pictures backgrounds. For example, you can add a gray solid color background or modify a picture to one of your choice.

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Configuring Your System

For Polycom VVX 1500 phones:

• You can select the pictures or designs displayed on the background. The supported formats include JPEG, BMP, and PNG and the maximum size is 800x480. A default picture is displayed when the phone starts up the first time.

Users can select which background appears on their individual phones. Users can also select a background from an image displayed by the digital picture frame feature (refer to Digital Picture Frame on page 4-39).

Note

Support for resolutions greater than 800x480 is inconsistent. Content may be truncated or nor displayed. Progressive/multiscan JPEG images are not supported at this time.

Configuration changes can be performed centrally at the provisioning server or locally:

Central (provisioning

server) Local Local Phone User

Configuration template:

features.cfg

Interface

To modify the backgrounds displayed on the supported SoundPoint IP phones:

1. Modify the features.cfg configuration file as follows:

a Open features.cfg in an XML editor.

b Locate the background parameter.

c For the solid backgrounds, set the name and RGB values. For example:

Specify which background will be displayed.

• Refer to <bg/> on page A-16.

On the Polycom VVX 1500, the user can save one of the images as the background by selecting Save as Background on the touch screen.

bg.hiRes.gray.pat.solid.3.name=”Gray” bg.hiRes.gray.pat.solid.3.red=”128” bg.hiRes.gray.pat.solid.3.green=”128” bg.hiRes.gray.pat.solid.3.blue=”128”

d For images, select a filename. For example:

bg.hiRes.gray.bm.3.name=”polycom.jpg” bg.hiRes.gray.bm.3.em.name=”polycomEM.jpg” bg.hiRes.gray.bm.3.adj=”0”

The default size for images on a phone is 320 x 160. The default size for images on an Expansion Module is 160 x 320. Use a photo editor on a computer to adjust the image you want to display. (Edit the image so the main subject is centered in the upper right corner of the display.)

Download the file to the provisioning server.

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e Save the modified features.cfg configuration file.

Automatic Off-Hook Call Placement

The phone supports an optional automatic off-hook call placement feature for each registration. This feature is sometimes referred to as ‘hot-dialing’.

Configuration changes can be performed centrally at the provisioning server:

Central (provisioning

server)

Call Hold

Central (provisioning

server)

Configuration template:

reg-advanced.cfg

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. The call hold reminder is always played through the speakerphone.

As of SIP 3.1, you can supply a Music on Hold URI if supported by the call server. For more information, refer to draft RFC draft-worley-service-example.

Configuration changes can be performed centrally at the provisioning server or locally:

Configuration template:

sip-interop.cfg

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

• Refer to <autoOffHook/> on page A-28.

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

• Refer to <SIP/> on page A-147. Specify local hold reminder options.

• Refer to <hold/><localReminder/> on page A-27. Specify the Music on Hold URI.

• Refer to <musicOnHold/> on page A-157.

Local Web Server

(if enabled)

Local Phone User Interface

4 - 20

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>-web.cfg on the provisioning server. Changes will permanently override global settings unless deleted through the

Reset Web Configuration menu selection and the <Ethernet address>-web.cfg is removed from the provisioning server.

Use the Call Server Configuration menu to specify whether or not to use RFC 2543 (c=0.0.0.0) outgoing hold signaling. The alternative is RFC 3264 (a=sendonly or a=inactive).

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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 be performed centrally at the provisioning server:

Central (provisioning

server)

Configuration template:

sip-interop.cfg

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

• Refer to <SIP/> on page A-147. Specify whether a transfer is blind or not.

• Refer to <call/> on page A-21.

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.

All phones support three-party local conferencing. The SoundPoint IP 450, 550, 560, 650, and 670 phones may support four-way local conferencing.

Note

Four-party conferencing requires a license key for activation. For more information, refer to Manage Conferences on page 4-22.

If theconference host of a three-party local conference ends the call, the other parties of the call may still be able to communicate. If the conference host of a four-party local conference ends the call, the conference ends.

The phone also supports centralized conferences for which external resources are used such as a conference bridge. This relies on network signaling.

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Configuration changes can be performed centrally at the provisioning server:

Central (provisioning

server)

Configuration template:

sip-interop.cfg

Note

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

• Refer to <call/> on page A-21. Specify whether or not the remaining parties can communicate after

the conference host exits the conference.

• Refer to <call/> on page A-21. Specify whether or not all parties hear sound effects while setting up a

conference.

• Refer to <call/> on page A-21. Specify which type of conference to establish and the address of the

centralized conference resource.

• Refer to <SIP/> on page A-147.

Manage Conferences

This feature is supported on the SoundPoint IP 450, 550, 560, 650, and 670 desktop phones, the SoundStation IP 7000 conference phone, and the Polycom VVX business media phone.

This feature requires a license key for activation on all phones except the SoundStation IP 7000 and the Polycom VVX 1500. Using this feature may require purchase of a license key or activation by Polycom channels. For more information, contact your Certified Polycom Reseller.

Central (provisioning

server)

Call Forward

4 - 22

The individual parties within a conference can be managed. New parties can be added and information about the conference participants can be viewed (for example, names, phone numbers, send/receive status or media flow, receive and transmit codecs, and hold status).

Configuration changes can be performed centrally at the provisioning server:

Configuration template:

features.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

Turn this feature on or off.

• Refer to <feature/> on page A-58.

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Configuring Your System

• When Do Not Disturb is active

• After an extended period of alerting

The user can elect to manually forward calls while they are in the alerting state to a predefined or manually specified destination. The call forwarding feature works in conjunction with the distinctive incoming call treatment feature (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 disables local Call Forward and DND features unless configured otherwise.

Server-based call forwarding will behave the same as per-SIP 2.1 feature with the following exception:

• 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).

Note

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

The Diversion field with a SIP header is often used by the call server to inform the phone of a call’s history. For example, when a phone has been set to enable call forwarding, the Diversion header allows the receiving phone to indicate who the call was from, and from which phone number it was forwarded. (For more information, refer to Header Support on page B-4.) .

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Configuration changes can be performed centrally at the provisioning server or locally:

Central (provisioning

server)

Local Web Server

Configuration template:

sip-interop.cfg

Configuration template:

reg-advanced.cfg

(if enabled)

Enable or disable server-based call forwarding.

• Refer to <SIP/> on page A-147. Enable or disable local call forwarding behavior when server-based

enabled.

• Refer to <SIP/> on page A-147. Enable or disable display of Diversion header and the order in which

to display the caller ID and number.

• Refer to <SIP/> on page A-147. 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.

• Refer to <divert/> on page A-48.

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

• Refer to <reg/>on page A-82.

Set all call diversion settings. Navigate to: http://<phoneIPAddress>/reg.htm Changes are saved to local flash and backed up to <Ethernet

address>-web.cfg on the provisioning server. Changes will permanently override global settings unless deleted through the

Reset Web Configuration menu selection and the <Ethernet address>-web.cfg is removed from the provisioning server.

Local Phone User Interface

Directed Call Pick-Up

4 - 24

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 provisioning server. Changes will permanently override global settings unless deleted through the

Reset Local Configuration menu selection and the <Ethernet address>-phone.cfg is removed from the provisioning 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. With many SIP servers, directed call pick-up is implemented using a particular star code sequence. With some SIP servers, specific network signaling is used to implement this feature.

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Configuring Your System

Configuration changes can be performed centrally at the provisioning server:

Central (provisioning

server)

Configuration template:

features.cfg

Configuration template:

sip-interop.cfg

Group Call Pick-Up

Central (provisioning

server)

Configuration template:

features.cfg

Turn this feature on or off.

• Refer to <feature/> on page A-58.

Determine the type of directed call pickup.

• Refer to <call/> on page A-21. Determine the type of SIP header to include.

• Refer to <voIpProt/> on page A-141.

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. With many SIP servers, group call pick-up is implemented using a particular star code sequence. With some SIP servers, specific network signaling is used to implement this feature.

Configuration changes can be performed centrally at the provisioning server:

Turn this feature on or off.

• Refer to <feature/> on page A-58.

Call Park/Retrieve

Central (provisioning

server)

Configuration template:

features.cfg

Configuration template:

sip-interop.cfg

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. With many SIP servers, this feature is implemented using a particular star code sequence. With some SIP servers, specific network signaling is used to implement this feature.

Configuration changes can be performed centrally at the provisioning server:

Turn this feature on or off.

• Refer to <feature/> on page A-58.

Determine the type of call park and retrieval string.

• Refer to <call/> on page A-21.

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Administrator’s Guide for the Polycom UC Software

Last Call Return

The phone allows server-based last call return. This feature depends on support from a SIP server. With many SIP servers, this feature is implemented using a particular star code sequence. With some SIP servers, specific network signaling is used to implement this feature.

Configuration changes can be performed centrally at the provisioning server:

Central (provisioning

server)

Configuration template:

features.cfg

Configuration template:

sip-interop.cfg

Turn this feature on or off.

• Refer to <feature/> on page A-58.

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

• Refer to <call/> on page A-21.

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

• Customizable Fonts

• Instant Messaging

• Multilingual User Interface

4 - 26

• Downloadable Fonts

• Synthesized Call Progress Tones

• Browser and Microbrowser

• Real-Time Transport Protocol Ports

• Network Address Translation

• Corporate Directory

• CMA Directory

• Recording and Playback of Audio Calls

• Digital Picture Frame

• Enhanced Feature Keys

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Configuring Your System

• Configurable Soft Keys

• LCD Power Saving

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

• Shared Call Appearances

• Bridged Line Appearance

• Busy Lamp Field

• Voice Mail Integration

• Multiple Registrations

• SIP-B Automatic Call Distribution

• Feature Synchronized Automatic Call Distribution

• Server Redundancy

• Presence

• CMA Presence

• Microsoft Live Communications Server 2005 Integration

• Access URL in SIP Message

• Static DNS Cache

• Display of Warnings from SIP Headers

• Quick Setup of Polycom Phones

Configurable Feature Keys

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

Note

Since there is no Redial key on the SoundPoint IP 32x/33x phone, the redial function cannot be remapped.

SoundStation IP 6000 and 7000 keys cannot be remapped to behave as Speed Dial keys.

The rules for remapping of key functions are:

• The phone keys that have removable key caps can be mapped to the following:

— Any function that is implemented as a removable key cap on any of

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

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— A speed-dial

— An enhanced feature key operation

— Null

• 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 be performed centrally at the provisioning server:

Central (provisioning

server)

Configuration Template:

features.cfg

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

• Refer to <key/> on page A-63.

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

Feature Key Layouts on page C-11.

Multiple Line Keys per Registration

More than one Line Key can be allocated to a single registration (phone number or line) on SoundPoint IP and Polycom VVX 1500 phones. The number of Line Keys allocated per registration is configurable.

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Configuring Your System

Configuration changes can be performed centrally at the provisioning server or locally:

Central (provisioning

server) Local Web Server

Configuration template:

reg-advanced.cfg

(if enabled)

Local Phone User Interface

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).

Specify the number of line keys to assign per registration.

• Refer to <reg/> on page A-82.

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 provisioning server. Changes will permanently override global settings unless deleted through the

Reset Web Configuration menu selection and the <Ethernet address>-phone.cfg is removed from the provisioning server.

Specify the number of line keys to assign per registration using the Line Configuration menu. Either the Web Server or the provisioning server configuration files or the local phone user interface should be used to configure registrations, not a mixture of these options. When the Line Configuration menu is used, it is assumed that all registrations use the same server.

Central (provisioning

server)

Configuration changes can be performed centrally at the provisioning server or locally:

Configuration template:

reg-basic.cfg

Configuration template:

reg-advanced.cfg

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

• Refer to <call/> on page A-21.

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.

• Refer to <reg/> on page A-82.

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Administrator’s Guide for the Polycom UC Software

Local Web Server

(if enabled)

Local Phone User Interface

Customizable Fonts

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 provisioning server. Changes will permanently override global settings unless deleted through the

Reset Web Configuration menu selection and the <Ethernet address>-phone.cfg is removed from the provisioning server.

Specify per-registration the number of calls that can be active or on hold per line key assigned to that registration using the Line Configuration menu. Either the Web Server or the provisioning server configuration files or the local phone user interface should be used to configure registrations, not a mixture of these options. When the Line Configuration menu is used, it is assumed that all registrations use the same server.

The phone’s user interface can be customized by changing the fonts 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.

Note

Central (provisioning server)

Configuration Template: region.cfg

Instant Messaging

Customizable fonts are not supported on the Polycom VVX 1500.

Configuration changes can be performed centrally at the provisioning server:

Specify fonts to overwrite existing ones or specify new fonts.

• Refer to <font/> on page A-61.

All phones (except the SoundPoint IP 32x/33x) support 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.

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Configuring Your System

Configuration changes can be performed centrally at the provisioning server:

Central (provisioning

server)

Configuration template:

features.cfg

Multilingual User Interface

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 higher resolution displays of the SoundPoint IP 450, 550, 560, 650, and 670, the SoundStation IP 5000, 6000, and 7000, and Polycom VVX 1500. A WGL4 character set is displayed by the SoundStation IP 7000. For more information, refer to

http://www.microsoft.com/OpenType/otspec/WGL4E.HTM.

For basic character support and extended character support (available on SoundPoint IP 450, 550, 560, 650 and 670 and SoundStation IP platforms), refer to <ml/> on page A-65. (Note that within a Unicode range, some characters may not be supported due to their infrequent usage.)

The SoundPoint IP and SoundStation IP user interface is available in the following languages by default: Simplified Chinese (if displayable), Danish, Dutch, English, French, German, Italian, Japanese (if displayable), Korean (if displayable), Norwegian, Polish, Brazilian Portuguese, Russian, Slovenian, International Spanish, and Swedish.

Turn this feature on or off.

• Refer to <feature/> on page A-58.

Note

The multilingual feature relies on dictionary files resident on the provisioning server. The dictionary files are downloaded from the provisioning 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 Polycom UC Software. The BootROM application is available in English only.

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Administrator’s Guide for the Polycom UC Software

Configuration changes can be performed centrally at the provisioning server or locally:

Central (provisioning

server)

Local Local Phone User

Configuration file:

site.cfg

Interface

Downloadable Fonts

New fonts can be loaded onto the phone. For guidelines on downloading fonts, refer to <font/> on page A-61.

Note

Downloadable fonts are not supported on the SoundStation IP 6000 and 7000 and the Polycom VVX 1500.

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

• Refer to <ml/> on page A-65.

languages, refer to To add new languages to those included with

the distribution: on page A-66.

The user can select the preferred language under the Settings menu. The languages appears in the list in the language itself. For example, German appears in the list as “Deutsch” and Swedish appears as “Svenska”. For administrator convenience, the ISO representation of each language is also included in the language selection menu.

Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the provisioning server. Changes will permanently override global settings unless deleted through the

Reset Local Configuration menu selection and the <Ethernet address>-phone.cfg is removed from the provisioning server.

For instructions on adding new

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 configurable for compatibility with worldwide telephony standards or local preferences. The built-in call progress tones are based on North American standard tones. For other geographies, certain tones may be reconfigured by the administrator using configuration files.

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Configuring Your System

Configuration changes can be performed centrally at the provisioning server:

Central (provisioning

server)

Configuration template: site.cfg

Configuration template: region.cfg

Browser and Microbrowser

The SoundPoint IP 450, 550, 560, 650, and 670 phones, the SoundStation IP 5000, 6000, and 7000 phones support an XHTML Microbrowser. This can be launched by pressing the Applications key or it can be accessed through the Menu key by selecting Applications.

Note

On some older phones, the Applications key is labeled Services.

The Polycom VVX 1500 phones running SIP 3.2.2 or later support a full browser. This can be launched by pressing the App key or it can accessed through the Menu key by selecting Applications.

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

• Refer to <chord/> on page A-119. Specify downloaded sampled audio files for advanced call progress

tones.

• Refer to <saf/> on page A-93.

Specify patterns.

• Refer to <pat/> on page A-96.

Note

If the browser uses over 30MB of memory and either the amount of free memory on the phone is below 6MB or the real time is between 1am to 5am, then the browser will restart. Once the browser has restarted, the last displayed web page is restored.

Two instances of the Microbrowser or Browser 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. (On the VVX 1500 phone, the idle browser allows interactivity to start up the active browser only.)

For more information, refer to the Web Application Developer’s Guide, which can be found at http://www.polycom.com/voicedocumentation/.

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Configuration changes can be performed centrally at the provisioning server or locally:

Central (provisioning

server)

Local Web Server

Configuration template:

applications.cfg

(if enabled)

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

• Refer to <mb/> on page A-69. Specify the telephone notification and state polling events to be

recorded and location of the push server.

• Refer to <apps/> on page A-10.

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>-web.cfg on the provisioning server. Changes will permanently override global settings unless deleted through the

Reset Web Configuration menu selection and the <Ethernet address>-web.cfg is removed from the provisioning 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.

4 - 34

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 fix the destination transport port to a specified value regardless of the negotiated port. This can be useful for communicating 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.

Poly SoundPoint IP 550 SoundPoint IP, SoundStation IP and Polycom VVX Administrator’s Guide, Version 3.3.0

Specifications and Main Features

Frequently Asked Questions

User Manual

Administrator’s Guide for the Polycom® UC Software

Contents

Introducing the Polycom UC Software Family of Phones

SoundPoint IP Desktop Phones

SoundStation IP Conference Phones

VVX 1500 Business Media Phone

Key Features of Your Polycom Phones

Overview

Where Polycom Phones Fit

Polycom Phone Software Architecture

BootROM

Polycom UC Software

Configuration

Resource Files

Available Features

New Features in Polycom UC Software 3.3.0

Setting up Your System

Setting Up the Network

DHCP or Manual TCP/IP Setup

Supported Provisioning Protocols

Modifying the Network Configuration

Setting Up the Provisioning Server

Deploying Phones From the Provisioning Server

Upgrading Polycom UC Software

Supporting Current SoundPoint IP, SoundStation IP, and VVX Phones

Supporting Legacy SoundPoint IP and SoundStation IP Phones

Provisioning SoundStation IP 7000 Phones Using C-Link

Provisioning VVX 1500 Phones Using a Polycom CMA System

Configuring Your System

Setting Up Basic Features

Call Log

Call Timer

Call Waiting

Called Party Identification

Calling Party Identification

Missed Call Notification

Connected Party Identification

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 Image Display

Ethernet Switch

Graphic Display Backgrounds

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

Setting Up Advanced Features

Configurable Feature Keys

Multiple Line Keys per Registration

Multiple Call Appearances

Customizable Fonts

Instant Messaging

Multilingual User Interface

Downloadable Fonts

Synthesized Call Progress Tones

Browser and Microbrowser

Real-Time Transport Protocol Ports