Polycom SOUNDPOINT SIP 3.1 User Manual

5.51 Mb

Administrator’s Guide for the Polycom® SoundPoint® IP/SoundStation® IP Family

SIP 3.1

August, 2008 Edition

1725-11530-310 Rev. A

SIP 3.1

Trademark Information

Polycom®, the Polycom logo design, SoundPoint® IP, SoundStation®, SoundStation VTX 1000®, ViaVideo®, ViewStation®, and Vortex® are registered trademarks of Polycom, Inc. Conference Composer™, Global Management System™, ImageShare™, Instructor RP™, iPower™, MGC™, PathNavigator™, People+Content™, PowerCam™, Pro-Motion™, QSX™, ReadiManager™, Siren™, StereoSurround™, V2IU™, Visual Concert™, VS4000™, VSX™, and the industrial design of SoundStation are trademarks of Polycom, Inc. in the United States and various other countries. All other trademarks are the property of their respective owners.

Patent Information

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

Disclaimer

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

Copyright Notice

Portions of the software contained in this product are:

Copyright © 1998, 1999, 2000 Thai Open Source Software Center Ltd. and Clark Cooper Copyright © 1998 by the Massachusetts Institute of Technology

Copyright © 1998-2003 The OpenSSL Project

Copyright © 1995-1998 Eric Young (eay@cryptsoft.com). All rights reserved Copyright © 1995-2002 Jean-Loup Gailly and Mark Adler

Copyright © 1996 - 2004, 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.

© 2008 Polycom, Inc. All rights reserved.

Polycom Inc. 4750 Willow Road

Pleasanton, CA 94588-2708 USA

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

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

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

About This Guide

The Administrator’s Guide for the SoundPoint IP / SoundStation IP family is for administrators who need to configure, customize, manage, and troubleshoot SoundPoint IP / SoundStation IP phone systems. This guide covers the SoundPoint IP 301, 320, 330, 430, 501, 550, 560, 600, 601, 650, and 670 desktop phones, and the SoundStation IP 4000 , 6000, and 7000 conference phones.

The following related documents for SoundPoint IP / SoundStation IP family are available:

Quick Start Guides, which describe how to assemble the phones

Quick User Guides, which describe the most basic features available on the phones

User Guides, which describe the basic and advanced features available on the phones

Developer’s Guide, which assists in the development of applications that run on the SoundPoint IP / SoundStation IP phone’s Microbrowser

Technical Bulletins, which describe workarounds to existing issues

Release Notes, which describe the new and changed features and fixed problems in the latest version of the software

For support or service, please contact your Polycom® reseller or go to Polycom Technical Support at http://www.polycom.com/support/voice/.

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

SoundPoint IP / SoundStation IP models: ___________________________

BootROM version: ________________________________________________

SIP Application version: ___________________________________________

Partner Platform: _________________________________________________

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Contents

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

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

SoundPoint IP Desktop Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1–1 SoundStation IP Conference Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1–4 Key Features of Your SoundPoint IP / SoundStation IP Phones . . . . . . . 1–6

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

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

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

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

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

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

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

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

New Features in SIP 3.1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2–13

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

Setting Up the Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–2 DHCP or Manual TCP/IP Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–2 Supported Provisioning Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–4 Modifying the Network Configuration . . . . . . . . . . . . . . . . . . . . . . . . . 3–5 Setting Up the Boot Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–12 Deploying Phones From the Boot Server . . . . . . . . . . . . . . . . . . . . . . . . . . 3–14 Upgrading SIP Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3–19 Supporting SoundPoint IP and SoundStation IP Phones . . . . . . . . . 3–19 Supporting SoundPoint IP 300 and 500 Phones . . . . . . . . . . . . . . . . . 3–20

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

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

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

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Call Timer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–3 Call Waiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–3 Called Party Identification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–4 Calling Party Identification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–4 Missed Call Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–4 Connected Party Identification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–5 Context Sensitive Volume Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–5 Customizable Audio Sound Effects . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–5 Message Waiting Indication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–6 Distinctive Incoming Call Treatment . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–6 Distinctive Ringing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–7 Distinctive Call Waiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–7 Do Not Disturb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–8 Handset, Headset, and Speakerphone . . . . . . . . . . . . . . . . . . . . . . . . . 4–8 Local Contact Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–9 Local Digit Map . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–12 Microphone Mute . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–13 Soft Key Activated User Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–13 Speed Dial . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–13 Time and Date Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–14 Idle Display Animation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–15 Ethernet Switch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–15 Graphic Display Backgrounds . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–16 Automatic Off-Hook Call Placement . . . . . . . . . . . . . . . . . . . . . . . . . . 4–17 Call Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–17 Call Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–18 Local / Centralized Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–19 Call Forward . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–20 Directed Call Pick-Up . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–21 Group Call Pick-Up . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–22 Call Park/Retrieve . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–22 Last Call Return . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–22

Setting Up Advanced Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–22 Configurable Feature Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–24 Multiple Line Keys per Registration . . . . . . . . . . . . . . . . . . . . . . . . . . 4–25 Multiple Call Appearances . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–25 Shared Call Appearances . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–26 Bridged Line Appearance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–27 Busy Lamp Field . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–28 Customizable Fonts and Indicators . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–29

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Instant Messaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–30 Multilingual User Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–30 Downloadable Fonts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–31 Synthesized Call Progress Tones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–32 Microbrowser . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–32 Real-Time Transport Protocol Ports . . . . . . . . . . . . . . . . . . . . . . . . . . 4–33 Network Address Translation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–34 Corporate Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–34 Recording and Playback of Audio Calls . . . . . . . . . . . . . . . . . . . . . . . 4–37 Daisy-Chaining Phones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–38 Provisioning Phones Over CLink . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–39 Enhanced Feature Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–40 Configurable Soft Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–50 Voice Mail Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–54 Multiple Registrations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–55 Automatic Call Distribution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–56 Server Redundancy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–56 Presence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–60 Microsoft Live Communications Server 2005 Integration . . . . . . . . 4–61 Access URL in SIP Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–65 Static DNS Cache . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–68 Display of Warnings from SIP Headers . . . . . . . . . . . . . . . . . . . . . . . 4–72

Setting Up Audio Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–73 Low-Delay Audio Packet Transmission . . . . . . . . . . . . . . . . . . . . . . . 4–74 Jitter Buffer and Packet Error Concealment . . . . . . . . . . . . . . . . . . . . 4–74 Voice Activity Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–74 DTMF Tone Generation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–75 DTMF Event RTP Payload . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–75 Acoustic Echo Cancellation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–75 Audio Codecs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–76 Background Noise Suppression . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–77 Comfort Noise Fill . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–77 Automatic Gain Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–78 IP Type-of-Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–78 IEEE 802.1p/Q . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–78 Voice Quality Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–79 Dynamic Noise Reduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–80 Treble/Bass Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–80

Setting Up Security Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–80 Local User and Administrator Privilege Levels . . . . . . . . . . . . . . . . . 4–81

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Custom Certificates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–81

Incoming Signaling Validation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–82

Secure Real-Time Transport Protocol . . . . . . . . . . . . . . . . . . . . . . . . . 4–82

Configuration File Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4–82

Configuring SoundPoint IP / SoundStation IP Phones Locally . . . . . . . 4–83

5Troubleshooting Your SoundPoint IP / SoundStation IP Phones . 5–1

Error Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–2 BootROM Error Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–2 Application Error Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–3 Status Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–4 Log Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–5 Reading a Boot Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–8 Reading an Application Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–9 Testing Phone Hardware . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–9 Power and Startup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–10 Controls . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–11 Access to Screens and Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–12 Calling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–13 Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–14 Audio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–15 Upgrading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5–15

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

Master Configuration Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–2 Application Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–4 Protocol <voIpProt/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–6 Dial Plan <dialplan/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–17 Localization <lcl/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–21 User Preferences <up/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–25 Tones <tones/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–27 Sampled Audio for Sound Effects <saf/> . . . . . . . . . . . . . . . . . . . . . A–30 Sound Effects <se/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–31 Voice Settings <voice/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–37 Quality of Service <QOS/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–55 Basic TCP/IP <TCP_IP/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–58 Web Server <httpd/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–63 Call Handling Configuration <call/> . . . . . . . . . . . . . . . . . . . . . . . . A–64

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Directory <dir/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–68 Presence <pres/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–72 Fonts <font/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–72 Keys <key/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–75 Backgrounds <bg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–77 Bitmaps <bitmap/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–80 Indicators <ind/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–80 Event Logging <log/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–84 Security <sec/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–88 License <license/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–89 Provisioning <prov/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–90 RAM Disk <ramdisk/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–90 Request <request/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–91 Feature <feature/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–92 Resource <res/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–93 Microbrowser <mb/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–95 Applications <apps/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–98 Peer Networking <pnet/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–100 DNS Cache <dns/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–100 Soft Keys <softkey/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–103

Per-Phone Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–106 Registration <reg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–107 Calls <call/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–111 Diversion <divert/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–114 Dial Plan <dialplan/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–116 Messaging <msg/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–119 Network Address Translation <nat/> . . . . . . . . . . . . . . . . . . . . . . A–120 Attendant <attendant/> . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–121 Roaming Buddies <roaming_buddies/> . . . . . . . . . . . . . . . . . . . . A–122 Roaming Privacy <roaming_privacy/> . . . . . . . . . . . . . . . . . . . . . A–123 User Preferences <user_preferences/> . . . . . . . . . . . . . . . . . . . . . . A–123

Flash Parameter Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A–124

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

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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–5 Adding a Background Logo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–6 BootROM/SIP Application Dependencies . . . . . . . . . . . . . . . . . . . . . . . . C–9 Migration Dependencies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–9 Multiple Key Combinations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–10 Default Feature Key Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–12 Internal Key Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–19 Assigning a VLAN ID Using DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–23 Parsing Vendor ID Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–24 Product, Model, and Part Number Mapping . . . . . . . . . . . . . . . . . . . . . C–26 Disabling PC Ethernet Port . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C–27

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

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

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1

Introducing the SoundPoint IP /

SoundStation IP Family

This chapter introduces the SoundPoint IP / SoundStation IP family, which is supported by the software described in this guide.

The SoundPoint IP / SoundStation IP family provides a powerful, yet flexible IP communications solution for Ethernet TCP/IP networks, delivering excellent voice quality. The high-resolution graphic display supplies content for call information, multiple languages, directory access, and system status. The SoundPoint IP / SoundStation IP family supports advanced functionality, including multiple call and flexible line appearances, HTTPS secure provisioning, presence, custom ring tones, and local conferencing.

The SoundPoint IP / SoundStation IP phones are end points in the overall network topology designed to interoperate with other compatible equipment including application servers, media servers, internet-working gateways, voice bridges, and other end points

The following models are described:

SoundPoint IP Desktop Phones

SoundStation IP Conference Phones

For a list of key features available on the SoundPoint IP / SoundStation IP phones running the latest software, refer to Key Features of Your SoundPoint IP / SoundStation IP 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/support/voice/. Additional options are also available. For more information, contact your Polycom distributor.

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The currently supported desktop phones are:

SoundPoint IP 301

SoundPoint IP 320/330

SoundPoint IP 430

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Introducing the SoundPoint IP / SoundStation IP Family

SoundPoint IP 501

SoundPoint IP 550/560

SoundPoint IP 600/601

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SoundPoint IP 650

SoundPoint IP 670

SoundStation IP Conference Phones

This section describes the current SoundPoint IP conference phones. For individual guides, refer to the product literature available at http://www.polycom.com/support/voice/. Additional options are also available. For more information, contact your Polycom distributor.

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The currently supported conference phones are:

SoundStation IP 4000

SoundStation IP 6000

SoundStation IP 7000

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Key Features of Your SoundPoint IP / SoundStation IP

Phones

The key features of the SoundPoint IP / SoundStation IP phones are:

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

Permits natural, high-quality, two-way conversations (one-way, monitor speaker in the SoundPoint IP 301)

Uses Polycom’s industry leading Acoustic Clarity Technology

Easy-to-use

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

Up to 18 dedicated hard keys for access to commonly used features

Up to four context-sensitive soft keys for further menu-driven activities

Platform independent

Supports multiple protocols and platforms enabling standardization on one phone for multiple locations, systems and vendors

Polycom’s support of the leading protocols and industry partners makes it a future-proof choice

Field upgradeable

Upgrade SoundPoint IP / SoundStation IP as standards develop and protocols evolve

Extends the life of the phone to protect your investment

Application flexibility for call management and new telephony applications

Large LCD

Easy-to-use, easily readable and intuitive interface

Support of rich application content, including multiple call appearances, presence and instant messaging, and XML services

4 line x 20 character monochrome LCD for the SoundPoint IP 301

102 x 23 pixel graphical LCD for the SoundPoint IP 320/330

160 x 80 pixel graphical grayscale LCD for the SoundPoint IP 501

320 x 160 pixel graphical grayscale LCD for the SoundPoint IP 550/560/600/601/650/670 (supports Asian characters)

248 x 68 pixel graphical LCD for the SoundStation IP 4000/6000

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

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Introducing the SoundPoint IP / SoundStation IP Family

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

Leverages existing infrastructure investment

No re-wiring with existing CAT 5 cabling

Simplifies installation

Power over Ethernet (PoE) port

Unused pairs on Ethernet port pairs are used to deliver power to the phone via a wall adapter allowing fewer wires to desktop

Optional accessory cable for CiscoR Inline Powering and IEEE 802.3af on the SoundPoint IP 301 and SoundPoint IP 501

Built-in PoE on the SoundPoint IP 550, 560, 600, 601, 650, and 670 and the SoundStation IP 6000 and 7000 (auto-sensing)

Multiple language support

Set on-screen language to your preference. Select from Chinese, Danish, Dutch, English, French, German, Italian, Japanese, Korean, Norwegian, Polish, Portuguese, Russian, Slovenian, Spanish, and Swedish

Note

In SIP 3.0, default support for Chinese, Japanese, and Korean was removed from

 

the SoundPoint IP 600 and 601.

 

 

Microbrowser

Supports a subset of XHTML constructs; otherwise runs like any other Web browser.

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Overview

This chapter provides an overview of the Session Initiation Protocol (SIP) application and how the phones fit into the network configuration.

SIP is the Internet Engineering Task Force (IETF) standard for multimedia conferencing over IP. It is an ASCII-based, application-layer control protocol (defined in RFC 3261) that can be used to establish, maintain, and terminate calls between two or more endpoints. Like other voice over IP (VoIP) protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call.

For the SoundPoint IP / SoundStation IP phones to successfully operate as a SIP endpoint in your network, it must meet the following requirements:

A working IP network is established.

Routers are configured for VoIP.

VoIP gateways are configured for SIP.

The latest (or compatible) SoundPoint IP / SoundStation IP phone SIP application image is available.

A call server is active and configured to receive and send SIP messages.

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

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

This chapter contains information on:

Where SoundPoint IP / SoundStation IP Phones Fit

Session Initiation Protocol Application Architecture

Available Features

New Features in SIP 3.1

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

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

SoundStation IP phones on the network, refer to Troubleshooting Your

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

Where SoundPoint IP / SoundStation IP Phones Fit

The phones connect physically to a standard office twisted-pair (IEEE 802.3) 10/100 megabytes per second Ethernet LAN and send and receive all data using the same packet-based technology. Since the phone is a data terminal, digitized audio being just another type of data from its perspective, the phone is capable of vastly more than traditional business phones. AsSoundPoint IP / SoundStation IP phones run the same protocols as your office personal computer, many innovative applications can be developed without resorting to specialized technology.

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Overview

Session Initiation Protocol Application Architecture

The software architecture of SIP application is made of 4 basic components:

BootROM—loads first when the phone is powered on

Application—software that makes the device a phone

Configuration—configuration parameters stored in separate files

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

Configuration

Resource

Files

bootROM

Application

BootROM

The bootROM is a small application that resides in the flash memory on the phone. All phones come from the factory with a bootROM pre-loaded.

The bootROM performs the following tasks in order:

1.Performs a power on self test (POST).

2.(Optional) Allows you to enter the setup menu where various network on provisioning options can be set.

The bootROM software controls the user interface when the setup menu is accessed.

3.Requests IP settings and accesses the boot server to look for any updates to the bootROM application.

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

4.If a new bootROM is downloaded, format the file system clearing out any application software or configuration files that may have been present.

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5.Download the master configuration file.

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

6.Examine the master configuration file for the name of the application file, and then look for this file on the boot server.

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

Note

If the Application is any SIP version prior to 1.5, the bootROM will also download all

 

the configuration files that are listed in the master configuration file.

 

 

7.Extract the application from flash memory.

8.Install the application into RAM, then upload a log file with events from the boot cycle.

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

Application

The application manages the VoIP stack, the digital signal processor (DSP), the user interface, and the network interaction. The application managed everything to do with the phone’s operation.

The application is a single file binary image and, as of SIP 1.5, contains a digital signature to prevent tampering or loading or rogue software images.

Warning If your phones are using bootROM 3.0 or later, the application must be signed.

All SIP 1.5 applications and later are signed, but later patched versions of 1.3 and 1.4 support this feature. Refer to the latest Release Notes to verify if the image is signed.

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

The application performs the following tasks in order:

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

These files are called sip.cfg and phone1.cfg by default. You can customized the filenames.

Note

If the Application is any SIP version prior to 1.5, the bootROM would have

 

downloaded all the configuration files that are listed in the master configuration file.

 

 

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Overview

2.Controls all aspects of the phone after it has restarted.

3.Uploads log files.

BootROM and Application Wrapper

Both the bootROM and the application run on multiple platforms (meaning all previously released versions of hardware that are still supported).

The file stored on the boot server is a wrapper, with multiple hardware specific images contained within. When a new bootROM or application is being saved, the file is read until a header matching the hardware model and revision are found, and then only this image is saved to flash memory.

Configuration

The SoundPoint IP / SoundStation IP phones can be configured automatically through files stored on a central boot server, manually through the phone’s local UI or web interface, or a combination of the automatic and manual methods.

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

The phone configuration files consist of:

Master Configuration Files

Application Configuration Files

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

Master Configuration Files

The master configuration files can be one of:

Specified master configuration file

Per-phone master configuration file

Default master configuration file

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

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Application Configuration Files

Typically, the files are arranged in the following manner although parameters may be moved around within the files and the filenames themselves can be changed as needed. These files dictate the behavior of the phone once it is running the executable specified in the master configuration file.

The application files are:

Application—It contains parameters that affect the basic operation of the phone such as voice codecs, gains, and tones and the IP address of an application server. All phones in an installation usually share this category of files. Polycom recommends that you create another file with your organization’s modifications. If you must change any Polycom templates, back them up first. By default, sip.cfg is included.

Per-phone—It contains parameters unique to a particular phone user. Typical parameters include:

display name

unique addresses

Each phone in an installation usually has its own customized version of user files derived from Polycom templates. By default, phone1.cfg is included.

Central Provisioning

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

In the central provisioning method, there are two major classifications of configuration files:

System configuration files

Per-phone configuration files

Parameters can be stored in the files in any order and can be placed in any number of files. The default is to have 2 files, one for per-phone setting and one for system settings. The per-phone file is typically loaded first, and could contain system level parameters, letting you override that parameter for a given user. For example, it might be desirable to set the default CODEC for a remote user differently than for all the users who reside in the head office. By adding the CODEC settings to a particular user’s per-phone file, the values in the system file are ignored.

Note

Verify the order of the configuration files. Parameters in the configuration file loaded

 

first will overwrite those in later configuration files.

 

 

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Overview

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

Manual Configuration

When the manual configuration method is employed, any changes made are stored in a configuration override file. This file is stored on the phone, but a copy will also be uploaded to the central boot server if one is being used. When the phone boots, this file is loaded by the application after any centrally provisioned files have been read, and its settings will override those in the centrally provisioned files.

This can create a lot of confusion about where parameters are being set, and so it is best to avoid using the manual method unless you have good reason to do so.

Resource Files

In addition to the application and the configuration files, the phones may require resource files that are used by some of the advanced features. These files are optional, but if the particular feature is being employed, these files are required.

Some examples of resource files include:

Language dictionaries

Custom fonts

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Ring tones

Synthesized tones

Contact directories

Available Features

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

Basic User Features

Automatic Off-Hook Call Placement—Supports an optional automatic off-hook call placement feature for each registration.

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

Connected Party Identification—The identity of the party to which the user has connected is displayed and logged, if the name is provided by the call server.

Context Sensitive Volume Control—The volume of user interface sound effects, such as the ringer, and the receive volume of call audio is adjustable.

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Overview

Customizable Audio Sound Effects—Audio sound effects used for incoming call alerting and other indications are customizable.

Directed Call Pick-Up and Group Call Pick-Up—Calls to another phone can be picked up by dialing the extension of the other phone. Calls to another phone within a pre-defined group can be picked up without dialing the extension of the other phone.

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

Distinctive Incoming Call Treatment—The phone can automatically apply distinctive treatment to calls containing specific attributes.

Distinctive Ringing—The user can select the ring type for each line and the ring type for specific callers can be assigned in the contact directory.

Do Not Disturb—A do-not-disturb feature is available to temporarily stop all incoming call alerting.

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). The SoundPoint IP 320, 330, 430, 500, 501, 550, 560, 600, 601, and 650 and 670 phones and SoundStation IP 4000, 6000, and 7000 phones are full-duplex speakerphones. The SoundPoint IP 301 phone is a listen-only speakerphone.

Idle Display Animation—All phones except the SoundPoint IP 301 can display a customized animation on the idle display in addition to the time and date.

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

Local / Centralized Conferencing—The phone can conference together the local user with the remote parties of two independent calls and can support centralized conferences for which external resources are used such as a conference bridge. 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 boot server and edited locally.

Local Digit Map—The phone has a local digit map to automate the setup phase of number-only calls.

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

Microphone Mute—When the microphone mute feature is activated, visual feedback is provided.

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Missed Call Notification—The phone can display the number of calls missed since the user last looked at the Missed Calls list.

Soft Key Activated User Interface—The user interface makes extensive use of intuitive, context-sensitive soft key menus.

Speed Dial—The speed dial system allows calls to be placed quickly from dedicated keys as well as from a speed dial menu.

Time and Date Display—Time and date can be displayed in certain operating modes such as when the phone is idle and during a call.

Advanced Features

Automatic Call Distribution—Supports ACD agent available and unavailable and allows ACD login and logout. Requires call server support.

Bridged Line Appearance—Calls and lines on multiple phones can be logically related to each other. Requires call server support.

Busy Lamp Field—Allows monitoring the hook status and remote party information of users through the busy lamp field (BLF) LEDs and displays on an attendant console phone. Requires call server support.

Configurable Feature Keys—Certain key functions can be changed from the factory defaults.

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.

Customizable Fonts and Indicators—The phone’s user interface can be customized by changing the fonts and graphic icons used on the display and the LED indicator patterns.

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

Instant Messaging—Supports sending and receiving instant text messages.

Microbrowser—The SoundPoint IP 430, 501, 550, 560, 600, 601, 650, and 670 desktop phones and the SoundStation IP 4000, 6000, and 7000 conference phones support an XHTML microbrowser.

Microsoft Live Communications Server 2005 Integration—SoundPoint IP and SoundStation IP phones can used with Microsoft Live Communications Server 2005 and Microsoft Office Communicator to help improve business efficiency and increase productivity and to share ideas and information immediately with business contacts. Requires call server support.

Multilingual User Interface—All phones except SoundPoint IP 301 have multilingual user interfaces.

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Overview

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.

Multiple Registrations—SoundPoint IP desktop 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.

Real-Time Transport Protocol Ports—The phone treats all realtime 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 what supports the .wav format. This feature is part of the Productivity Suite.

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 the standard audio codecs.

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Administrator’s Guide SoundPoint IP / SoundStation IP

Automatic Gain Control—Designed for hands-free operation, boosts the transmit gain of the local user in certain circumstances.

Background Noise Suppression—Designed primarily for hands-free operation, reduces background noise to enhance communication in noisy environments.

Comfort Noise Fill—Designed to help provide a consistent noise level to the remote user of a hands-free call.

DTMF Event RTP Payload—Conforms to RFC 2833, which describes a standard RTP-compatible technique for conveying DTMF dialing and other telephony events over an RTP media stream.

DTMF Tone Generation—Generates dual tone multi-frequency (DTMF) tones in response to user dialing on the dial pad.

IEEE 802.1p/Q—The phone will tag all Ethernet packets it transmits with an 802.1Q VLAN header.

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

Jitter Buffer and Packet Error Concealment—Employs a high-performance jitter buffer and packet error concealment system designed to mitigate packet inter-arrival jitter and out-of-order or lost (lost or excessively delayed by the network) packets.

Low-Delay Audio Packet Transmission—Designed to minimize latency for audio packet transmission.

Voice Activity Detection—Conserves network bandwidth by detecting periods of relative “silence” in the transmit data path and replacing that silence efficiently with special packets that indicate silence is occurring.

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.

Security Features

Local User and Administrator Privilege Levels—Several local settings menus are protected with two privilege levels, user and administrator, each with its own password.

Configuration File Encryption—Confidential information stored in configuration files must be protected (encrypted). The phone can recognize encrypted files, which it downloads from the boot server and it can encrypt files before uploading them to the boot server.

Custom Certificates—When trying to establish a connection to a boot server for application provisioning, the phone trusts certificates issued by widely recognized certificate authorities (CAs).

Incoming Signaling Validation—Levels of security are provided for validating incoming network signaling.

2 - 12

Overview

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 SIP 3.1

Note

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 2.1.3.

 

Any new features introduced after SIP 2.1.3 are not supported.

 

 

The following new features were introduced in SIP 3.1:

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.

Configurable Soft Keys—Allows customers to create their own soft keys and have them displayed with or without the standard SoundPoint IP and SoundStation IP soft keys.

Enhanced Feature Keys—Allows customers to redefine soft keys to suit their needs. In SIP 3.0, this feature required a license key.

Dynamic Noise Reduction— Provides maximum microphone sensitivity, while automatically reducing background noise on SoundStation IP 7000 conference phones.

Treble/Bass Controls—Equalizes the tone of the high and low frequency sound from the speakers on SoundStation IP 7000 conference phones.

Display of Warnings from SIP Headers—Displays a “pop-up” to user that is found in the Warning Field from a SIP header.

The following existing features were changed in SIP 3.1:

Call Forward—The Diversion field can be used by the call server to inform the phone of a call’s history.

Call Hold—If supported by the call server, you can supply a Music on Hold URI.

Local Contact Directory—A new “Label” field has been added to each contact directory entry.

Busy Lamp Field—The attendant can now monitor all calls states and pickup remote calls.

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Administrator’s Guide SoundPoint IP / SoundStation IP

Microbrowser—An XML API allows for the creation of more advanced applications.

Multilingual User Interface—Polish and Slovenian are now available as languages choices.

Documentation of the newly released SoundPoint IP 560 and 670 desktop phones and SoundStation IP 6000 and 7000 conference phones has also been added.

2 - 14

3

Setting up Your System

Your SoundPoint IP / SoundStation IP SIP phone is designed to be used like a regular phone on a public switched telephone network (PSTN).

This chapter provides basic instructions for setting up your SoundPoint IP / SoundStation IP phones. This chapter contains information on:

Setting Up the Network

Setting Up the Boot Server

Deploying Phones From the Boot Server

Upgrading SIP Application

Because of the large number of optional installations and configurations that are available, this chapter focuses on one particular way that the SIP application and the required external systems might initially be installed and configured in your network.

For more information on configuring your system, refer to Configuring Your System on page 4-1. For more information on the configuration files required for setting up your system, refer to Configuration Files on page A-1.

For installation and maintenance of Polycom SoundPoint IP / SoundStation IP phones, the use of a boot server is strongly recommended. This allows for flexibility in installing, upgrading, maintaining, and configuring the phone. Configuration, log, and directory files are normally located on this server. Allowing the phone write access to the server is encouraged.

The phone is designed such that, if it cannot locate a boot server when it boots up, it will operate with internally saved parameters. This is useful for occasions when the boot server is not available, but is not intended to be used for long-term operation of the phones.

However, if you want to register a single SoundPoint IP / SoundStation IP phone, refer to “Quick Tip 44011: Registering Standalone SoundPoint IP and SoundStation IP Phones“ at http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_Technical_Bull etins_pub.html .

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Administrator’s Guide SoundPoint IP / SoundStation IP

Setting Up the Network

Regardless of whether or not you will be installing a centrally provisioned system, you must perform basic TCP/IP network setup, such as IP address and subnet mask configuration, to get your organization’s phones up and running.

The bootROM application uses the network to query the boot server for upgrades, which is an optional process that will happen automatically when properly deployed. For more information on the basic network settings, refer to DHCP or Manual TCP/IP Setup on page 3-2.

The bootROM on the phone performs the provisioning functions of downloading the bootROM, the <Ethernet address>.cfg file, and the SIP application, and uploading log files. For more information, refer to Supported Provisioning Protocols on page 3-4.

Basic network settings can be changed during bootROM download using the bootROM’s setup menu. A similar menu system is present in the application for changing the same network parameters. For more information, refer to Modifying the Network Configuration on page 3-5.

DHCP or Manual TCP/IP Setup

Basic network settings can be derived from DHCP, or entered manually using the phone’s LCD-based user interface, or downloaded from configuration files.

Polycom recommends using DHCP where possible to eliminate repetitive manual data entry.

The following table shows the manually entered networking parameters that may be overridden by parameters obtained from a DHCP server, an alternate DHCP server, or configuration file:

Parameter

DHCP Option

DHCP

Alternate

Configuration File

Local

DHCP

(application only)

FLASH

 

 

 

 

 

 

 

 

D priority when more than one source exists D

 

 

 

 

 

 

 

 

 

1

2

3

4

 

 

 

 

 

 

IP address

1

-

-

 

 

 

 

 

 

subnet mask

1

-

-

 

 

 

 

 

 

IP gateway

3

-

-

 

 

 

 

 

 

3 - 2

 

 

 

 

 

 

Setting up Your System

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Alternate

Configuration File

 

Local

 

Parameter

DHCP Option

DHCP

DHCP

(application only)

 

FLASH

 

 

Refer to DHCP

-

 

 

boot server

Menu on page

 

 

 

 

 

 

address

3-7

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

151

-

-

 

 

 

Note: This value

 

 

 

 

 

 

SIP server address

is configurable.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SNTP server

42 then 4

-

 

 

address

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SNTP GMT offset

2

-

 

 

 

 

 

 

 

 

 

 

DNS server IP

6

-

-

 

 

address

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

alternate DNS

6

-

-

 

 

server IP address

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

DNS domain

15

-

-

 

 

 

 

 

 

 

 

 

 

Refer to DHCP

Warning: Cisco Discovery Protocol (CDP) overrides Local FLASH

 

 

Menu on page

that overrides DHCP VLAN Discovery.

 

 

 

VLAN ID

3-7

 

 

 

 

 

 

 

 

 

 

 

 

 

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 in Time Synchronization <sntp/> on page

 

A-59.

 

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

 

the switch supports CDP.

 

 

 

In the case where you do not have control of your DHCP server or do not have

 

the ability to set the DHCP options, an alternate method of automatically

 

discovering the provisioning server address is required. Connecting to a

 

secondary DHCP server that responds to DHCP INFORM queries with a

 

requested boot server value is one possibility. For more information, refer to

 

http://www.ietf.org/rfc/rfc3361.txt?number=3361 and

 

http://www.ietf.org/rfc/rfc3925.txt?number=3925.

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Administrator’s Guide SoundPoint IP / SoundStation IP

Supported Provisioning Protocols

The bootROM performs the provisioning functions of downloading configuration files, uploading and downloading the configuration override file and user directory, and downloading the dictionary and uploading log files.

The protocol that will be used to transfer files from the boot server depends on several factors including the phone model and whether the bootROM or SIP application stage of provisioning is in progress. By default, the phones are shipped with FTP enabled as the provisioning protocol. If an unsupported protocol is specified, this may result in a defined behavior (see the table below for details of which protocol the phone will use). The Specified Protocol listed in the table can be selected in the Server Type field or the Server Address can include a transfer protocol, for example http://usr:pwd@server (refer to Server Menu on page 3-9). The boot server address can be an IP address, domain string name, or URL. The boot server address can also be obtained through DHCP. Configuration file names in the <Ethernet address>.cfg file can include a transfer protocol, for example https://usr:pwd@server/dir/file.cfg. If a user name and password are specified as part of the server address or file name, they will be used only if the server supports them.

Note

 

A URL should contain forward slashes instead of back slashes and should not

 

 

contain spaces. Escape characters are not supported. If a user name and

 

 

password are not specified, the Server User and Server Password will be used

 

 

(refer to Server Menu on page 3-9).

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Protocol used by

 

Protocol used by

 

 

 

 

bootROM

 

SIP Application

 

 

 

 

301, 320, 330, 430,

 

301, 320, 330, 430,

 

 

Specified

501, 550, 560, 600,

 

501, 550, 560, 600,

 

 

601, 650, 670, 4000,

 

601, 650, 670, 4000,

 

 

Protocol

6000, 7000

 

6000, 7000

 

 

FTP

FTP

 

FTP

 

 

 

 

 

 

 

 

TFTP

TFTP

 

TFTP

 

 

 

 

 

 

 

 

HTTP

HTTP

 

HTTP

 

 

 

 

 

 

 

 

HTTPS

HTTP

 

HTTPS

 

 

 

 

 

 

 

Note

 

 

 

 

 

 

 

There are two types of FTP methods—active and passive. As of SIP 1.5 (and

 

 

bootROM 3.0), the SIP application is no longer compatible with active FTP. At that

 

 

time, secure provisioning was implemented.

 

 

 

 

 

 

 

3 - 4

Setting up Your System

Note

Setting Option 66 to tftp://192.168.9.10 has the effect of forcing a TFTP download.

 

Using a TFTP URL (for example, tftp://provserver.polycom.com) has the same

 

effect.

 

 

 

For downloading the bootROM and application images to the phone, the

 

secure HTTPS protocol is not available. To guarantee software integrity, the

 

bootROM will only download cryptographically signed bootROM or

 

application images. For HTTPS, widely recognized certificate authorities are

 

trusted by the phone and custom certificates can be added (refer to Trusted

 

Certificate Authority List on page C-1).

Modifying the Network Configuration

You can access the network configuration menu:

During bootROM Phase. The network configuration menu is accessible during the auto-boot countdown of the bootROM phase of operation. Press the Setup soft key to launch the main menu.

During Application Phase. The network configuration menu is accessible from the phone’s main menu. Select Menu>Settings>Advanced>Admin Settings>Network Configuration. Advanced Settings are locked by default. Enter the administrator password to unlock. The factory default password is 456.

Phone network configuration parameters may be modified by means of:

Main Menu

DHCP Menu

Server Menu

Ethernet Menu

Syslog Menu

Use the soft keys, the arrow keys, the Select and Delete keys to make changes.

Certain parameters are read-only due to the value of other parameters. For example, if the DHCP Client parameter is enabled, the Phone IP Addr and Subnet Mask parameters are dimmed or not visible since these are guaranteed to be supplied by the DHCP server (mandatory DHCP parameters) and the statically assigned IP address and subnet mask will never be used in this configuration.

Resetting to Factory Defaults

The basic network configuration referred to in the subsequent sections can be reset to factory defaults using a multiple key combination described in Multiple Key Combinations on page C-10.

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Administrator’s Guide SoundPoint IP / SoundStation IP

Main Menu

The following configuration parameters can be modified on the main setup menu:

Name

Possible Values

Description

 

 

 

DHCP Client

Enabled, Disabled

If enabled, DHCP will be used to obtain the parameters

 

 

discussed in DHCP or Manual TCP/IP Setup on page

 

 

3-2.

 

 

 

DHCP Menu

 

Refer to DHCP Menu on page 3-7.

 

 

Note: Disabled when DHCP client is disabled.

 

 

 

Phone IP Address

dotted-decimal IP address

Phone’s IP address.

 

 

Note: Disabled when DHCP client is enabled.

 

 

 

Subnet Mask

dotted-decimal subnet

Phone’s subnet mask.

 

mask

Note: Disabled when DHCP client is enabled.

 

 

 

 

 

IP Gateway

dotted-decimal IP address

Phone’s default router.

 

 

 

Server Menu

 

Refer to Server Menu on page 3-9.

 

 

 

SNTP Address

dotted-decimal IP address

Simple Network Time Protocol (SNTP) server from

 

OR

which the phone will obtain the current time.

 

 

 

domain name string

 

 

 

 

GMT Offset

-13 through +12

Offset of the local time zone from Greenwich Mean

 

 

Time (GMT) in half hour increments.

 

 

 

DNS Server

dotted-decimal IP address

Primary server to which the phone directs Domain

 

 

Name System (DNS) queries.

 

 

 

DNS Alternate Server

dotted-decimal IP address

Secondary server to which the phone directs Domain

 

 

Name System queries.

 

 

 

DNS Domain

domain name string

Phone’s DNS domain.

 

 

 

Ethernet

 

Refer to Ethernet Menu on page 3-11.

 

 

 

EM Power

Enabled, Disabled

This parameter is relevant if the phone gets Power over

 

 

Ethernet (PoE). If enabled, the phone will set power

 

 

requirements in CDP to 12W so that up to three

 

 

Expansion Modules (EM) can be powered. If disabled,

 

 

the phone will set power requirements in CDP to 5W

 

 

which means no Expansion Modules can be powered (it

 

 

will not work).

 

 

 

Syslog

 

Refer to Syslog Menu on page 3-11.

 

 

 

3 - 6

 

 

 

Setting up Your System

 

 

 

 

Note

 

 

 

 

A parameter value of “???” indicates that the parameter has not yet been set and

 

 

saved in the phone’s configuration. Any such parameter should have its value set

 

 

before continuing.

 

 

 

The EM Power parameter is only available on SoundPoint IP 601 and 650 phones.

Note

 

 

 

 

 

 

 

To switch the text entry mode on the SoundPoint IP 330/320, press the #. You may

 

 

want to use URL or IP address modes when entering server addresses.

 

 

 

 

 

 

DHCP Menu

 

 

 

The DHCP menu is accessible only when the DHCP client is enabled. The

 

 

following DHCP configuration parameters can be modified on the DHCP

 

 

menu:

 

 

 

 

 

Possible

 

Name

Values

Description

Timeout

1 through 600

Number of seconds the phone waits for secondary DHCP Offer

 

 

 

messages before selecting an offer.

 

 

 

Boot Server

0=Option 66

The phone will look for option number 66 (string type) in the

 

 

 

response received from the DHCP server. The DHCP server

 

 

 

should send address information in option 66 that matches one

 

 

 

of the formats described for Server Address in the 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.

 

 

 

- 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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Administrator’s Guide SoundPoint IP / SoundStation IP

 

Possible

 

Name

Values

Description

 

 

 

Boot Server (continued)

1=Custom

The phone will look for the option number specified by the Boot

 

 

Server Option parameter (below), and the type specified by

 

 

the Boot Server Option Type parameter (below) in the

 

 

response received from the DHCP server.

 

 

If the DHCP server sends nothing, the 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

The phone will first use the custom option if present or use

 

66

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.

 

 

 

Boot Server Option

128 through 254

When the boot server parameter is set to Custom, this

 

(Cannot be the

parameter specifies the DHCP option number in which the

 

same as VLAN ID

phone will look for its boot server.

 

Option)

 

 

 

 

Boot Server Option Type

0=IP Address,

When the Boot Server parameter is set to Custom, this

 

1=String

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.

 

 

 

3 - 8

 

 

 

 

 

 

Setting up Your System

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Possible

 

 

Name

 

Values

Description

VLAN Discovery

 

0=Disabled

No VLAN discovery through DHCP.

 

 

 

(default)

 

 

 

 

 

 

 

 

 

 

 

1=Fixed

Use predefined DHCP vendor-specific option values of 128,

 

 

 

 

 

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

 

 

 

 

 

be ignored

 

 

 

 

 

 

 

 

 

2=Custom

Use the number specified in the VLAN ID Option field as the

 

 

 

 

 

DHCP private option value.

 

 

 

 

 

 

VLAN ID Option

 

128 through 254

The DHCP private option value (when VLAN Discovery is set

 

 

 

(Cannot be the

to Custom).

 

 

 

same as Boot

For more information, refer to Assigning a VLAN ID Using

 

 

 

Server Option)

DHCP on page C-23.

 

 

 

(default is 129)

 

 

 

 

 

 

 

 

 

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.

 

 

 

 

 

 

 

 

 

 

 

Server Menu

 

 

 

 

 

 

The following server configuration parameters can be modified on the Server

 

 

 

 

menu:

 

 

 

 

 

 

 

Name

 

Possible Values

 

Description

 

 

 

 

 

Server Type

 

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

 

The protocol that the phone will use to obtain

 

 

3=HTTPS, 4=FTPS, 5=Invalid

configuration and phone application files from the boot

 

 

 

 

 

 

server. Refer to Supported Provisioning Protocols on

 

 

 

 

 

 

page 3-4.

 

 

 

 

 

 

Note: Active FTP is not supported for bootROM version

 

 

 

 

 

 

3.0 or later. Passive FTP is still supported.

 

 

 

 

 

 

Note: Only implicit FTPS is supported.

 

 

 

 

 

 

 

3 - 9

Administrator’s Guide SoundPoint IP / SoundStation IP

Name

 

Possible Values

Description

 

 

 

 

Server Address

 

dotted-decimal IP address

The boot server to use if the DHCP client is disabled, the

 

 

OR

DHCP server does not send a boot server option, or the

 

 

domain name string

Boot Server parameter is set to Static. The phone can

 

 

OR

contact multiple IP addresses per DNS name. These

 

 

URL

redundant boot servers must all use the same protocol. If

 

 

All addresses can be followed

a URL is used it can include a user name and password.

 

 

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

 

 

by an optional directory and

 

 

directory and the master configuration file can be

 

 

optional file name.

 

 

specified.

 

 

 

 

 

 

 

 

Note: ":", "@", or "/" can be used in the user name or

 

 

 

 

password these characters if they are correctly escaped

 

 

 

 

using the method specified in RFC 1738.

 

 

 

 

Server User

 

any string

The user name used when the phone logs into the server

 

 

 

 

(if required) for the selected Server Type.

 

 

 

 

Note: If the Server Address is a URL with a user name,

 

 

 

 

this will be ignored.

 

 

 

 

Server Password

 

any string

The password used when the phone logs in to the server

 

 

 

 

if required for the selected Server Type.

 

 

 

 

Note: If the Server Address is a URL with user name and

 

 

 

 

password, this will be ignored.

 

 

 

 

File Transmit Tries

 

1 to 10

The number of attempts to transfer a file. (An attempt is

 

 

Default 3

defined as trying to download the file from all IP

 

 

 

 

addresses that map to a particular domain name.)

 

 

 

 

Retry Wait

 

0 to 300

The minimum amount of time that must elapse before

 

 

Default 1

retrying a file transfer, in seconds. The time is measured

 

 

 

 

from the start of a transfer attempt which is defined as the

 

 

 

 

set of upload/download transactions made with the IP

 

 

 

 

addresses that map to a given boot server's DNS host

 

 

 

 

name. If the set of transactions in an attempt is equal to or

 

 

 

 

greater than the Retry Wait value, then there will be no

 

 

 

 

further delay before the next attempt is started.

 

 

 

 

For more information, refer to Deploying Phones From the

 

 

 

 

Boot Server on page 3-14.

 

 

 

 

Network

 

Cable/DSL,

The network environment the phone is operating in.

 

 

LAN,

The default value is Cable/DSL.

 

 

Dial-up

 

 

 

 

 

 

 

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.

 

 

 

 

 

3 - 10

 

 

 

 

Setting up Your System

 

 

 

 

 

 

 

 

Ethernet Menu

 

 

 

 

The following Ethernet configuration parameters can be modified on the

 

 

 

Ethernet menu:

 

 

 

 

 

Name

 

Possible Values

Description

 

 

 

 

CDP Compatibility

 

Enabled, Disabled

If enabled, the phone will use a CDP compatibility

 

 

 

 

method. It also reports PoE power usage to the switch.

 

 

 

 

The default value is Enabled.

 

 

 

 

VLAN ID

 

Null, 0 through 4094

Phone’s 802.1Q VLAN identifier. The default value is Null.

 

 

 

 

Note: Null = no VLAN tagging

 

 

 

 

VLAN Filtering

 

Enabled, Disabled

Filter received Ethernet packets so that the TCP/IP stack

 

 

 

 

does not process bad data or too much data.

 

 

 

 

Enable/disable the VLAN filtering state.

 

 

 

 

The default value is Disabled.

 

 

 

 

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

The network speed over the Ethernet.

 

 

1

= 10HD

The default value is Auto.

 

 

2

= 10FD

 

 

HD means half duplex and FD means full duplex.

 

 

3

= 100HD

 

 

Note: Polycom recommends that you do not change this

 

 

4

= 100FD

 

 

5

= 1000FD

setting.

 

 

 

 

PC Port Mode

 

0 = Auto

The network speed over the Ethernet.

 

 

1

= 10HD

The default value is Auto.

 

 

2

= 10FD

 

 

HD means half duplex and FD means full duplex.

 

 

3

= 100HD

 

 

Note: Polycom recommends that you do not change this

 

 

4

= 100FD

 

 

5

= 1000FD

setting unless you want to disable the PC port.

 

 

-1 = Disabled

 

 

 

 

 

 

Note

 

The LAN Port Mode and PC Port Mode parameters are only available on

 

 

 

SoundPoint IP 330, 430, 550, 560, 601, 650, and 670 phones.

Only the SoundPoint IP 560 and 670 supports the LAN Port Mode and PC Port

Mode setting of 1000FD.

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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Administrator’s Guide SoundPoint IP / SoundStation IP

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

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

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

The following syslog configuration parameters can be modified on the Syslog menu:

Name

Possible Values

Description

 

 

 

Server Address

dotted-decimal IP address

The syslog server IP address or host name.

 

OR

The default value is NULL.

 

domain name string

 

 

 

 

 

Server Type

None=0,

The protocol that the phone will use to write to the syslog

 

UDP=1,

server.

 

TCP=2,

If set to “None”, transmission is turned off, but the server

 

TLS=3

 

address is preserved.

 

 

 

 

 

Facility

0 to 23

A description of what generated the log message. For

 

 

more information, refer to section 4.1.1 of RFC 3164.

 

 

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

 

 

 

Render Level

0 to 6

Specifies the lowest class of event that will be rendered to

 

 

syslog. It is based on log.render.level and can be a

 

 

lower value.

 

 

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

 

 

on page A-86.

 

 

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

 

 

 

Prepend MAC

Enabled, Disabled

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

Address

 

log message sent to the syslog server.

 

 

 

Setting Up the Boot Server

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

Multiple boot servers can be configured by having the boot server DNS name map to multiple IP addresses. The default number of boot servers is one and the maximum number is eight. The following protocols are supported for redundant boot servers: HTTPS, HTTP, and FTP. For more information on the protocol used on each platform, refer to Supported Provisioning Protocols on page 3-4.

3 - 12

Setting up Your System

 

All of the boot servers must be reachable by the same protocol and the content

 

available on them must be identical. The parameters described in section

 

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

 

server will be tried for a file transfer and also how long to wait between each

 

attempt. The maximum number of servers to be tried is configurable. For more

 

information, contact your Certified Polycom Reseller.

Note

 

Be aware of how logs, overrides and directories are uploaded to servers that maps

 

to multiple IP addresses. The server that these files are uploaded to may change

 

over time.

 

If you want to use redundancy for uploads, synchronize the files between servers in

 

the background.

 

However, you may want to disable the redundancy for uploads by specifying

 

specific IP addresses instead of URLs for logs, overrides, and directory in the

 

<MAC-address>.cfg .

 

 

To set up the boot server:

Note

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

 

 

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

Polycom recommends that you use RFC-compliant servers.

 

2. Create account and home directory.

Note

 

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.

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.

 

 

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Administrator’s Guide SoundPoint IP / SoundStation IP

3.Copy all files from the distribution zip file to the phone home directory. Maintain the same folder hierarchy.

The distribution zip file contains:

sip.ld (including a separate one for every supported model)

sip.cfg

phone1.cfg

000000000000.cfg

000000000000-directory~.xml

SoundPointIP-dictionary.xml (one of each supported language)

SoundPointIPWelcome.wav

Refer to the Release Notes for a detailed description of each file in the distribution.

Boot Server Security Policy

You must decide on a boot server security policy.

Polycom recommends allowing file uploads to the boot server where the security environment permits. This allows event log files to be uploaded and changes made by the phone user to the configuration (through the web server and local user interface) and changes made to the directory to be backed up.

For organizational purposes, configuring a separate log file directory is recommended, but not required. (For more information on LOG_FILE_DIRECTORY, refer to Master Configuration Files on page A-2.)

File permissions should give the minimum access required and the account used should have no other rights on the server.

The phone's server account needs to be able to add files to which it can write in the log file directory and the root directory. It must also be able to list files in all directories mentioned in the <MAC-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 Boot Server

You can successfully deploy SoundPoint IP and SoundStation IP phones from one or more boot servers.

3 - 14

Setting up Your System

Multiple boot servers can be configured by having the boot server DNS name map to multiple IP addresses. The default number of boot servers is one and the maximum number is eight. HTTPS, HTTP, and FTP are supported for redundant boot servers.

For all SoundPoint IP and SoundStation IP phones, follow the normal provisioning process in the next section, Provisioning Phones. However, if you have decided to daisy-chain two SoundStation IP 7000 conference phones together, read the information in Provisioning SoundStation IP 7000 Phones Using CLink on page 3-18 to understand the different provisioning options available.

Provisioning Phones

To deploy phones from the boot server:

Note

For more information on encrypting configuration files, refer to Encrypting

 

Configuration Files on page C-4.

 

 

1.(Optional) Create per-phone configuration files by performing the following steps:

Note

This step may be omitted if per-phone configuration is not needed.

 

 

aObtain a list of phone Ethernet addresses (barcoded label on underside of phone and on the outside of the box).

bCreate per-phone phone[MACaddress].cfg file by using the phone1.cfg file from the distribution as templates.

For more information on the phone1.cfg file, refer to Per-Phone Configuration on page A-106.

Note

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

 

interchangeable.

 

 

cEdit contents of phone[MACaddress].cfg if desired. For example, edit the parameters.

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Administrator’s Guide SoundPoint IP / SoundStation IP

 

2. (Optional) Create new configuration file(s) in the style of sip.cfg by

 

performing the following steps:

Note

 

For more information on why to create another configuration file, refer to the

 

“Configuration File Management on SoundPoint IP Phones” whitepaper at

 

www.polycom.com/support/voice/ .

 

 

 

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

 

<SIP/> on page A-10.

 

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

 

Configuration on page A-4.

 

Most of the default settings are typically adequate, however, if SNTP

 

settings are not available through DHCP, the SNTP GMT offset and

 

(possibly) the SNTP server address will need to be edited for the correct

 

local conditions. Changing the default daylight savings parameters will

 

likely be necessary outside of North American locations.

 

a (Optional) Disable the local web (HTTP) server or change its

 

signalling port if local security policy dictates.

 

b Change the default location settings for user interface language and

 

time and date format.

 

3. (Optional) Create a master configuration file by performing the following

 

steps:

 

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

 

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

 

For more information, refer to Master Configuration Files on page

 

A-2.

 

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

 

so that it references the appropriate phone[MACaddress].cfg file.

 

For example, replace the reference to phone1.cfg with

 

phone[MACaddress].cfg.

3 - 16

Setting up Your System

cEdit the CONFIG_FILES attribute of the <Ethernet address>.cfg files so that it references the appropriate sipXXXX.cfg file.

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

dEdit the LOG_FILE_DIRECTORY attribute of the <Ethernet address>.cfg files so that it points to the log file directory.

eEdit the CONTACT_DIRECTORY attribute of the <Ethernet address>.cfg files so that it points to the organization’s contact directory.

4.Reboot the phones by pressing the reboot multiple key combination. For more information, refer to Multiple Key Combinations on page C-10.

The bootROM and SIP application modify the APPLICATION APP_FILE_PATH attribute of the <Ethernet address>.cfg files so that it references the appropriate sip.ld files.

For example, the reference to sip.ld is changed to 2345-11605-001.sip.ld to boot the SoundPoint IP 601 image.

Note

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-24.

 

 

5.Monitor the boot server event log and the uploaded event log files (if permitted).

Ensure that the configuration process completed correctly. All configuration files used by the boot server are logged.

You can now instruct your users to start making calls.

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Administrator’s Guide SoundPoint IP / SoundStation IP

Provisioning SoundStation IP 7000 Phones Using CLink

Normally the SoundStation IP family conference phone is provisioned over the Ethernet by the boot server. However, when two SoundStation IP family 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

5

12-foot Ethernet Cable

Interconnect Cable

25-foot

 

Network Cable

4

The provisioning over CLink feature is automatically enabled when a SoundStation IP family phone is not connected to the Ethernet. Both SoundStation IP family phones must be running the same version of the SIP application.

The steps for provisioning the secondary SoundStation IP family phone are the same as for the primary SoundStation IP family phone. You can reboot the primary without rebooting the secondary. However, the primary and secondary should be rebooted together for the primary/secondary relationship to be recognized. If you power up both SoundStation IP family phones, the primary will power up first.

Currently, provisioning over CLink is supported for the following configurations of SoundStation IP family conference phones:

Two SoundStation IP family conference phone daisy-chained together

Two SoundStation IP family conference phone daisy-chained together with one external microphone, specifically designed for the SoundStation IP family conference phone

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

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

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

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

3 - 18

Setting up Your System

Upgrading SIP Application

You can upgrade the SIP application that is running on the SoundPoint IP and SoundStation IP phones in your organization. The exact steps that you perform are dependent on the version of the SIP application that is currently running on the phones and the version that want to upgrade to.

The bootROM, application executable, and configuration files can be updated automatically through the centralized provisioning model. These files are read-only by default.

Most organization can use the instructions shown in the next section, Supporting SoundPoint IP and SoundStation IP Phones.

However, if your organization has a mixture of SoundPoint IP 300 and/or 500 phones deployed along with other models, you will need to change the phone configuration files to continue to support the SoundPoint IP 300 and IP 500 phones when software releases SIP 2.2.0 or later are deployed. These models were discontinued as of May 2006. In this case, refer to Supporting SoundPoint IP 300 and 500 Phones on page 3-20.

Warning The SoundPoint IP 300 and 500 phones will be supported on the latest maintenance patch release of the SIP 2.1 software stream—currently SIP 2.1.3. 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 for these changes to be effective.

Supporting SoundPoint IP and SoundStation IP Phones

To automatically update:

1.Back up old application and configuration files.

The old configuration can be easily restored by reverting to the backup files.

2.Customize new configuration files or apply new or changed parameters to the old configuration files.

Differences between old and new versions of configuration files are explained in the Release Notes that accompany the software. Both mandatory and optional changes may present. Changes to site-wide configuration files such as sip.cfg can be done manually, but a scripting tool is useful to change per-phone configuration files.

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Administrator’s Guide SoundPoint IP / SoundStation IP

Warning The configuration files listed in CONFIG_FILES attribute of the master configuration file must be updated when the software is updated. Any new configuration files must be added to the CONFIG_FILES attribute in the appropriate order.

Mandatory changes must be made or the software may not behave as expected.

For more information, refer to the “Configuration File Management on SoundPoint IP Phones” whitepaper at www.polycom.com/support/voice/ .

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

4.Reboot the phones by pressing the reboot multiple key combination. For more information, refer to Multiple Key Combinations on page C-10.

Since the APPLICATION APP_FILE_PATH attribute of the <Ethernet address>.cfg files references the individual sip.ld files, it is possible to verify that an update is applied to phones of a particular model.

For example, the reference to sip.ld is changed to 2345-11605-001.sip.ld to boot the SoundPoint IP 601 image.

The phones can be rebooted remotely through the SIP signaling protocol. Refer to Special Events <specialEvent/> on page A-16.

The phones can be configured to periodically poll the boot server to check for changed configuration files or application executable. If a change is detected, the phone will reboot to download the change. Refer to Provisioning <prov/> on page A-90.

Supporting SoundPoint IP 300 and 500 Phones

With enhancements in BootROM 4.0.0 and SIP 2.1.2, you can modify the

000000000000.cfg or <Ethernet address>.cfg configuration file to direct phones to load the software image and configuration files based on the phone model number. Refer to Master Configuration Files on page A-2.

The SIP 2.2.0 or later software distributions contain only the new distribution files for the new release. You must rename the sip.ld, sip.cfg, and phone1.cfg from a previous 2.1.2 distribution that is compatible with SoundPoint IP 300 and 500 phones.

The following procedure must be used for upgrading to SIP 2.2.0 or later for installations that have SoundPoint IP 300 and 500 phones deployed. It is also recommended that this same approach be followed even if SoundPoint IP 300 and 500 phones are not part of the deployment as it will simplify management of phone systems with future software releases.

3 - 20

Setting up Your System

To upgrade your SIP application:

1.Do one of the following steps:

a Place the bootrom.ld file corresponding to BootROM revision 4.0.0 (or later) onto the boot server.

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

2.Copy sip.ld, sip.cfg and phone1.cfg from the SIP2.2.0 or later release distribution onto the boot server.

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

3.Rename sip.ld, sip.cfg, and phone1.cfg from the previous distribution to sip_212.ld, sip_212.cfg, and phone1_212.cfg respectively on the boot server.

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

4.Modify the 000000000000.cfg file, if required, to match your configuration file structure.

For example:

<APPLICATION APP_FILE_PATH="sip.ld" APP_FILE_PATH_SPIP500="sip_212.ld" APP_FILE_PATH_SPIP300="sip_212.ld"

CONFIG_FILES="[PHONE_MAC_ADDRESS]-user.cfg, phone1.cfg, sip.cfg" CONFIG_FILES_SPIP500="[PHONE_MAC_ADDRESS]-user.cfg,

phone1_212.cfg, sip_212.cfg" CONFIG_FILES_SPIP300="[PHONE_MAC_ADDRESS]-user.cfg,

phone1_212.cfg, sip_212.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="" OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY="" />

5.Remove any <Ethernet address>.cfg files that may have been used with earlier releases from the boot server.

Note

This approach takes advantage of an enhancement that was added in

 

SIP2.0.1/BootROM 3.2.1 that allows for the substitution of the phone specific

 

[MACADDRESS] inside configuration files. This avoids the need to create unique

 

<Ethernet address>.cfg files for each phone such that the default

 

000000000000.cfg file can be used for all phones in a deployment.

 

If this approach is not used, then changes will need to be made to all the <Ethernet

 

address>.cfg files for SoundPoint IP 300 and 500 phones or all of the <Ethernet

 

address>.cfg files if it is not explicitly known which phones are SoundPoint IP 300

 

and 500 phones.

 

 

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Administrator’s Guide SoundPoint IP / SoundStation IP

For more information, refer to “Technical Bulletin 35311: Supporting SoundPoint IP 300 and IP 500 Phones with SIP 2.2 and Later Releases“ at http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_T echnical_Bulletins_pub.html .

3 - 22

4

Configuring Your System

After you set up your SoundPoint IP / SoundStation IP phones on the network, you can allow users to place and answer calls using the default configuration, however, you may be require some basic changes to optimize your system for best results.

This chapter provides information for making configuration changes for:

Setting Up Basic Features

Setting Up Advanced Features

Setting Up Audio Features

Setting Up Security Features

This chapter also provides instructions on:

Configuring SoundPoint IP / SoundStation IP Phones Locally

To troubleshoot any problems with your SoundPoint IP / SoundStation IP phones on the network, refer to Troubleshooting Your SoundPoint IP / SoundStation IP Phones on page 5-1. For more information on the configuration files, refer to Configuration Files on page A-1.

Setting Up Basic Features

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

Call Log

Call Timer

Call Waiting

Called Party Identification

Calling Party Identification

Missed Call Notification

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Administrator’s Guide SoundPoint IP / SoundStation IP

Connected Party Identification

Context Sensitive Volume Control

Customizable Audio Sound Effects

Message Waiting Indication

Distinctive Incoming Call Treatment

Distinctive Ringing

Distinctive Call Waiting

Do Not Disturb

Handset, Headset, and Speakerphone

Local Contact Directory

Local Digit Map

Microphone Mute

Soft Key Activated User Interface

Speed Dial

Time and Date Display

Idle Display Animation

Ethernet Switch

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

4 - 2

Configuring Your System

Call Log

The phone maintains a call log. The log contains call information such as remote party identification, time and date, and call duration. It can be used to redial previous outgoing calls, return incoming calls, and save contact information from call log entries to the contact directory.

The call log is stored in volatile memory and is maintained automatically by the phone in three separate lists: Missed Calls, Received Calls and Placed Calls. The call lists can be cleared manually by the user and will be erased when the phone is restarted.

 

Note

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

 

 

 

list. Missed calls appear as

 

and received calls appear as

 

.

 

 

 

 

 

 

 

 

 

 

 

 

The “call list” feature can be disabled on all SoundPoint IP and SoundStation IP

 

 

 

platforms except the SoundPoint IP 330/320 and SoundStation IP 7000.

 

 

 

 

 

 

 

 

 

 

Configuration changes can performed centrally at the boot server:

 

 

 

 

 

 

Central

 

Configuration File:

Enable or disable all call lists or individual call lists.

 

 

(boot server)

 

sip.cfg

 

For more information, refer to Feature <feature/> on page A-92.

 

 

 

 

 

 

 

 

 

Call Timer

A call timer is provided on the display. A separate call timer is maintained for each distinct call in progress. The call duration appears in hours, minutes, and seconds.

There are no related configuration changes.

Call Waiting

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

Configuration changes can performed centrally at the boot server:

Central

Configuration File:

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

(boot server)

phone1.cfg

active.

 

 

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

 

 

A-113.

 

 

 

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

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Administrator’s Guide SoundPoint IP / SoundStation IP

Called Party Identification

The phone displays and logs the identity of the remote party specified for outgoing calls. This is the party that the user intends to connect with.

There are no related configuration changes.

Calling Party Identification

The phone displays the caller identity, derived from the network signalling, when an incoming call is presented, if the information is provided by the call server. For calls from parties for which a directory entry exists, the local name assigned to the directory entry may optionally be substituted.

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

Central

Configuration File:

Specify whether or not to use directory name substitution.

(boot server)

sip.cfg

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

 

 

A-25.

 

 

 

Local

Web Server

Specify whether or not to use directory name substitution.

 

(if enabled)

Navigate to: http://<phoneIPAddress>/coreConf.htm#us

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

Missed Call Notification

The phone can display the number of calls missed since the user last looked at the Missed Calls list. The types of calls that are counted as “missed” can be configured per registration. Remote missed call notification can be used to notify the phone when a call originally destined for it is diverted by another entity such as a Session Initiation Protocol (SIP) server.

Note

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

 

list.

 

 

4 - 4

 

 

Configuring Your System

 

 

 

 

Configuration changes can performed centrally at the boot server:

 

 

 

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

For more information, refer to Feature <feature/> on page A-92.

 

 

 

 

Configuration file:

Specify per-registration whether all missed-call events or only

 

phone1.cfg

remote/server-generated missed-call events will be displayed.

 

 

For more information, refer to Missed Call Configuration

 

 

<serverMissedCall/> on page A-112.

 

 

 

Connected Party Identification

The identity of the remote party to which the user has connected is displayed and logged, if the name and ID is provided by the call server. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion.

There are no related configuration changes.

Context Sensitive Volume Control

The volume of user interface sound effects, such as the ringer, and the receive volume of call audio is adjustable. While transmit levels are fixed according to the TIA/EIA-810-A standard, receive volume is adjustable. For SoundPoint IP and phones, if using the default configuration parameters, the receive handset/headset volume resets to nominal after each call to comply with regulatory requirements. Handsfree volume persists with subsequent calls.

Configuration changes can performed centrally at the boot server:

Central

Configuration file:

Adjust receive and handset/headset volume.

(boot server)

sip.cfg

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

 

 

page A-42.

 

 

 

Customizable Audio Sound Effects

Audio sound effects used for incoming call alerting and other indications are customizable. Sound effects can be composed of patterns of synthesized tones or sample audio files. The default sample audio files may be replaced with alternates in .wav file format. Supported .wav formats include:

mono G.711 (13-bit dynamic range, 8-khz sample rate)

mono L16/16000 (16-bit dynamic range, 16-kHz sample rate)

mono L16/32000 (16-bit dynamic range, 32-kHz sample rate)

mono L16/48000 (16-bit dynamic range, 48-kHz sample rate)

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Administrator’s Guide SoundPoint IP / SoundStation IP

 

Note

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

 

 

 

phones. L16/32000 and L16/48000 are only supported on SoundPoint IP 7000

 

 

 

phones.

 

 

Note

 

 

 

 

 

 

The alternate sampled audio sound effect files must be present on the boot server

 

 

 

or the Internet for downloading at boot time.

 

 

 

 

 

 

 

 

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

 

 

 

 

Central

 

Configuration File:

Specify patterns used for sound effects and the individual tones or

(boot server)

 

sip.cfg

 

sampled audio files used within them.

 

 

 

 

For more information, refer to Sampled Audio for Sound Effects

 

 

 

 

<saf/> on page A-30 or Sound Effects <se/> on page A-31.

 

 

 

 

Local

 

Web Server

Specify sampled audio wave files to replace the built-in defaults.

 

 

(if enabled)

Navigate to http://<phoneIPAddress>/coreConf.htm#sa

 

 

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

 

 

override global settings unless deleted through the Reset Local

 

 

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

 

 

removed from the boot server.

 

 

 

Message Waiting Indication

 

 

 

 

The phone will flash a message-waiting indicator (MWI) LED when instant

 

 

 

messages and voice messages are waiting.

 

 

 

Configuration changes can performed centrally at the boot server:

 

 

 

 

Central

 

Configuration file:

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

(boot server)

 

phone1.cfg

For more information, refer to Message Waiting Indicator <mwi/>

 

 

 

 

on page A-120.

 

 

 

 

Specify whether MWI notification is displayed for registration x

 

 

 

 

(pre-SIP 2.1 behavior is enabled).

 

 

 

 

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

 

 

 

 

A-25.

 

 

 

 

 

Distinctive Incoming Call Treatment

The phone can automatically apply distinctive treatment to calls containing specific attributes. The distinctive treatment that can be applied includes customizable alerting sound effects and automatic call diversion or rejection. Call attributes that can trigger distinctive treatment include the calling party name or SIP contact (number or URL format).

For related configuration changes, refer to Local Contact Directory on page 4-9.

4 - 6

Configuring Your System

Distinctive Ringing

There are three options for distinctive ringing:

1.The user can select the ring type for each line. This option has the lowest priority.

2.The ring type for specific callers can be assigned in the contact directory. For more information, refer to Distinctive Incoming Call Treatment, the previous section. This option has a higher priority than option 1 and a lower priority than option 3.

3.The voIpProt.SIP.alertInfo.x.value and voIpProt.SIP.alertInfo.x.class fields can be used to map calls to specific ring types. This option has the highest priority.

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

Central

Configuration file:

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

(boot server)

sip.cfg

For more information, refer to Alert Information <alertInfo/> on

 

 

 

page A-15.

 

 

 

 

Configuration file:

Specify the ring type to be used for each line.

 

phone1.cfg

For more information, refer to Registration <reg/> on page A-107.

 

 

 

 

XML File: <Ethernet

This file can be created manually using an XML editor.

 

address>-directory.

For more information, refer to Local Contact Directory on page

 

xml

 

4-9.

 

 

 

Local

Local Phone User

The user can edit the ring types selected for each line under the

 

Interface

Settings menu. The user can also edit the directory contents.

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

 

Distinctive Call Waiting

The voIpProt.SIP.alertInfo.x.value and voIpProt.SIP.alertInfo.x.class fields can be used to map calls to distinct call waiting types, currently limited to two styles.

Configuration changes can performed centrally at the boot server:

Central

Configuration file:

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

(boot server)

sip.cfg

For more information, refer to Alert Information <alertInfo/> on

 

 

 

page A-15.

 

 

 

 

4 - 7

Administrator’s Guide SoundPoint IP / SoundStation IP

Do Not Disturb

A Do Not Disturb (DND) feature is available to temporarily stop all incoming call alerting. Calls can optionally be treated as though the phone is busy while DND is enabled. DND can be configured as a per-registration feature.

Incoming calls received while DND is enabled are logged as missed. For more information on forwarding calls while DND is enabled, refer to Call Forward on page 4-20.

Server-based DND is active if the feature is enabled on both the phone and the server and the phone is registered. The server-based DND feature is applicable for all registrations on the phone (no per-registration mode) and it disables local Call Forward and DND features.

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

There is no indication on the phone’s user interface whether or not server-based DND is active.

If server-based DND is enabled, but inactive, and the user presses the DND key or selects the DND option on the Feature menu, the “Do Not Disturb” message does not appear on the user’s phone (incoming call alerting will continue).

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

Central

Configuration file:

Enable or disable server-based DND.

(boot server)

sip.cfg

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

 

 

Specify whether or not DND results in incoming calls being given

 

 

busy treatment.

 

 

For more information, refer to Call Handling Configuration <call/>

 

 

on page A-64.

 

 

 

 

Configuration file:

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

 

phone1.cfg

For more information, refer to Registration <reg/>on page A-107.

 

 

Specify whether DND is treated as a per-registration feature or a

 

 

global feature on the phone.

 

 

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

 

 

A-116.

 

 

 

Local

Local Phone User

Enable or disable DND using the “Do Not Disturb” key on the

 

Interface

SoundPoint IP 301, 501, 550, 560, 600, 601, and 650 and 670 or the

 

 

“Do Not Disturb” option on the Features menu on the SoundPoint IP

 

 

320, 330, and 430 and SoundStation IP 4000, 6000, and 7000.

 

 

 

Handset, Headset, and Speakerphone

SoundPoint IP phones come standard with a handset and a dedicated connector is provided for a headset (not supplied). The SoundPoint IP 320, 330, 430, 500, 501, 550, 560, 600, 601, 650, and 670 desktop phones and SoundStation

4 - 8

Configuring Your System

IP 4000, 6000, and 7000 conference phones are full-duplex speakerphones. The SoundPoint IP 301 phones is a listen-only speakerphone. The SoundPoint IP phones provide dedicated keys for convenient selection of either the speakerphone or headset.

Only the SoundPoint IP 320, 330, 430, 550, 560, 650, and 670 desktop phones can be configured to use the electronic hookswitch. For more information, refer to “Technical Bulletin 35150: Using an Electronic Hookswitch with SountPoint IP Phones“at http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_T echnical_Bulletins_pub.html .

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

Central

Configuration file:

Enable or disable persistent headset mode.

(boot server)

sip.cfg

For more information, refer to User Preferences <up/> on page A-25.

 

 

Enable or disable hands-free speakerphone mode.

 

 

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

 

 

A-25.

 

 

 

 

Configuration file:

Specify whether or not the electronic hookswitch is enabled and what

 

phone1.cfg

type of headset is attached.

 

 

For more information, refer to User Preferences

 

 

<user_preferences/>on page A-107.

 

 

 

Local

Web Server

Enable or disable persistent headset mode.

 

(if enabled)

Navigate to: http://<phoneIPAddress>/coreConf.htm#us

 

 

 

 

Local Phone User

Enable or disable persistent headset mode through the Settings

 

Interface

menu (Settings > Basic > Preferences > Headset > Headet

 

 

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 boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

Local Contact Directory

The phone maintains a local contact directory. The directory can be downloaded from the boot server and edited locally (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.

4 - 9

Administrator’s Guide SoundPoint IP / SoundStation IP

 

Note

If a user makes a change to the local contact directory, there is a five second

 

 

 

timeout before it is uploaded to the boot server as <mac-address>-directory.cfg.

 

 

 

 

 

 

 

 

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

 

 

 

 

Central

 

Configuration file:

Set whether the directory uses volatile storage on the phone

(boot server)

 

sip.cfg

 

(required on the SoundPoint IP 500 platform for directories greater

 

 

 

 

than 25 entries).

 

 

 

 

For more information, refer to Local Directory <local/> on page

 

 

 

 

A-68.

 

 

 

 

Specify whether or not the local contact directory is read only.

 

 

 

 

For more information, refer to Local Directory <local/> on page

 

 

 

 

A-68.

 

 

 

 

 

 

 

XML file:

 

A sample file named 000000000000-directory~.xml (Note the extra

 

 

000000000000-direct

“~” in the filename) is included with the application file distribution.

 

 

This file can be used as a template for the per-phone <Ethernet

 

 

ory.xml

 

 

 

 

 

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

This file can be created manually using an XML editor.

 

 

address>-directory.

For information on file format, refer to the next section, Local Contact

 

 

xml

 

Directory File Format.

 

 

 

 

Local

 

Local Phone User

The user can edit the directory contents if configured in that way.

 

 

Interface

 

Changes will be stored in the phone’s flash file system and backed up

 

 

 

 

to the boot server copy of <Ethernet address>-directory.xml if this

 

 

 

 

is configured. When the phone boots, the boot server copy of the

 

 

 

 

directory, if present, will overwrite the local copy.

 

 

 

 

 

Local Contact Directory File Format

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

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

<item_list> <item>

<lb>Mr</lb>

<ln>Doe</ln>

<fn>John</fn>

4 - 10

Configuring Your System

<ct>1001</ct>

<sd>1</sd>

<rt>1</rt>

<dc/>

<ad>0</ad>

<ar>0</ar>

<bw>0</bw>

<bb>0</bb>

</item>

...

<item>

<lb>Dr</lb>

<ln>Smith</ln>

<fn>Bill</fn>

<ct>1003</ct>

<sd>3</sd>

<rt>3</rt>

<dc/>

<ad>0</ad>

<ar>0</ar>

<bw>0</bw>

<bb>0</bb>

</item> </item_list>

</directory>

Element

Permitted Values

Interpretation

 

 

 

lb

UTF-8 encoded string

label

 

of up to 40 bytes

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.

 

 

 

fn

UTF-8 encoded string

first name

 

of up to 40 bytes

Note: In some cases, this will be less than 40 characters due to

 

 

UTF-8’s variable length encoding.

 

 

 

ln

UTF-8 encoded string

last name

 

of up to 40 bytes

Note: In some cases, this will be less than 40 characters due to

 

 

UTF-8’s variable length encoding.

 

 

 

ct

UTF-8 encoded string

contact

 

containing digits (the

Used by the phone to address a remote party in the same way that a

 

user part of a SIP

 

string of digits or a SIP URL are dialed manually by the user. This

 

URL) or a string that

 

element is also used to associate incoming callers with a particular

 

constitutes a valid SIP

 

directory entry.

 

URL

 

Note: This field cannot be null or duplicated.

 

 

 

 

 

4 - 11

Administrator’s Guide SoundPoint IP / SoundStation IP

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 330/320 and the SoundStation IP 6000

 

 

and 7000, the maximum speed-dial index is 99.

 

 

 

rt

Null, 1 to 21

ring type

 

 

When incoming calls can be associated with a directory entry by

 

 

matching the address fields, this field is used to specify ring type to

 

 

be used.

 

 

 

dc

UTF-8 encoded string

divert contact

 

containing digits (the

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.

 

 

 

Local Digit Map

The phone has a local digit map feature to automate the setup phase of number-only calls. When properly configured, this feature eliminates the need for using the Dial or Send soft key when making outgoing calls. As soon as a digit pattern matching the digit map is found, the call setup process will complete automatically. The configuration syntax is 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 also possible to strip a trailing # from the digits sent or to replace certain matched digits (with the introduction of “R” to the digit map).

4 - 12

 

 

Configuring Your System

 

 

 

 

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

 

 

 

Central

Configuration file:

Specify impossible match behavior, trailing # behavior, digit map

(boot server)

sip.cfg

matching strings, and time out value.

 

 

For more information, refer to Dial Plan <dialplan/> on page A-17.

 

 

 

 

Configuration file:

Specify per-registration impossible match behavior, trailing #

 

phone1.cfg

behavior, digit map matching strings, and time out values that

 

 

override those in sip.cfg.

 

 

For more information, refer to Dial Plan <dialplan/> on page

 

 

A-116.

 

 

 

Local

Web Server

Specify impossible match behavior, trailing # behavior, digit map

 

(if enabled)

matching strings, and time out value.

 

Navigate to: http://<phoneIPAddress>/appConf.htm#ls

 

 

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

Microphone Mute

A microphone mute feature is provided. When activated, visual feedback is provided. This is a local function and cannot be overridden by the network.

There are no related configuration changes.

Soft Key Activated User Interface

The user interface makes extensive use of intuitive, context-sensitive soft key menus. The soft key function is shown above the key on the graphic display.

There are no related configuration changes.

Speed Dial

Entries in the local directory can be linked to the speed dial system. The speed dial system allows calls to be placed quickly from dedicated keys as well as from a speed dial menu.

For SoundPoint IP 320/330 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 SoundStation IP 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.

4 - 13

Administrator’s Guide SoundPoint IP / SoundStation IP

 

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

 

 

 

Central

XML file:

The <sd>x</sd> element in the <Ethernet address>-directory.xml

(boot server)

<Ethernet

file links a directory entry to a speed dial resource within the phone.

Speed dial entries are mapped automatically to unused line keys (line

 

address>-directory.

 

keys are not available on the SoundStation IP 4000, 6000 and 7000)

 

xml

 

and are available for selection within the speed dial menu. (Press the

 

 

 

 

up-arrow key from the idle display to jump to SpeedDial).

 

 

For more information, refer to Local Contact Directory on page

 

 

4-9.

 

 

 

Local

Local Phone User

The next available Speed Dial Index is assigned to new directory

 

Interface

entries. Key pad short cuts are available to facilitate assigning and

 

 

modifying the Speed Dial Index value for entries in the directory. The

 

 

Speed Dial Index field is used to link directory entries to speed dial

 

 

operations.

 

 

Changes will be stored in the phone’s flash file system and backed up

 

 

to the boot server copy of <Ethernet address>-directory.xml if this

 

 

is configured. When the phone boots, the boot server copy of the

 

 

directory, if present, will overwrite the local copy.

 

 

 

Time and Date Display

The phone maintains a local clock and calendar. Time and date can be displayed in certain operating modes such as when the phone is idle and during a call. The clock and calendar must be synchronized to a remote Simple Network Time Protocol (SNTP) timeserver. The time and date displayed on the phone will flash continuously until a successful SNTP response is received to indicate that they are not accurate. The time and date display can use one of several different formats and can be turned off.

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

Central

Configuration file:

Turn time and date display on or off.

(boot server)

sip.cfg

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

 

 

A-25.

 

 

Set the time and date display formats.

 

 

For more information, refer to Date and Time <datetime/> on

 

 

page A-25.

 

 

Set the basic SNTP settings and daylight savings parameters.

 

 

For more information, refer to Time Synchronization <sntp/> on

 

 

page A-59.

 

 

 

4 - 14

 

 

Configuring Your System

 

 

 

 

 

 

Local

Web Server

Set the basic SNTP and daylight savings settings.

 

(if enabled)

Navigate to: http://<phoneIPAddress>/coreConf.htm#ti

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

 

Local Phone User

The basic SNTP settings can be made in the Network Configuration

 

Interface

menu.

 

 

For more information, refer to DHCP or Manual TCP/IP Setup on

 

 

page 3-2.

 

 

The user can edit the time and date format and enable or disable the

 

 

time and date display under the Settings menu.

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. They will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection.

 

 

 

Idle Display Animation

 

 

All phones except the SoundPoint IP 301 can display a customized animation

 

on the idle display in addition to the time and date. For example, a company

 

logo could be displayed (refer to Adding a Background Logo on page C-6).

 

Configuration changes can performed centrally at the boot server:

 

 

 

Central

Configuration file:

To turn idle display animation on or off.

(boot server)

sip.cfg

For more information, refer to Indicators <ind/> on page A-80.

 

 

To replace the animation used for the idle display.

 

 

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

 

 

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

 

 

<IP_7000/> on page A-81.

 

 

To change the position of the idle display animation.

 

 

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

 

 

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

 

 

<IP_7000/> on page A-83.

 

 

 

Ethernet Switch

The SoundPoint IP phones contain two Ethernet ports, labeled LAN and PC, and an embedded Ethernet switch that runs at full line-rate. The 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

4 - 15

Administrator’s Guide SoundPoint IP / SoundStation IP

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 disabled the PC Ethernet port, refer to Disabling PC Ethernet Port on page C-27.

SoundPoint IP Switch - Port Priorities

To help ensure good voice quality, the Ethernet switch embedded in the SoundPoint IP phones should be configured to give voice traffic emanating from the phone higher transmit priority than those from a device connected to the PC port. If not using a VLAN (VLAN set to blank in the setup menu), this will automatically be the case. If using a VLAN, ensure that the 802.1p priorities for both default and real-time transport protocol (RTP) packet types are set to 2 or greater. Otherwise, these packets will compete equally with those from the PC port. For more information, refer to Quality of Service <QOS/> on page A-55.

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 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 grey solid color background or modify a picture to one

 

of your choice.

Note

 

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.

 

 

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

1.Modify the sip.cfg configuration file as follows: a Open sip.cfg in an XML editor.

b Locate the background parameter.

4 - 16

Configuring Your System

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

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 boot server.

e Save the modified sip.cfg configuration file. Configuration changes can performed centrally at the boot server:

Central

Configuration file:

Specify which background will be displayed.

(boot server)

phone1.cfg

For more information, refer to Backgrounds <bg/> on page A-77.

 

 

 

Automatic Off-Hook Call Placement

The phone supports an optional automatic off-hook call placement feature for each registration.

Configuration changes can performed centrally at the boot server:

Central

Configuration file:

Specify which registrations have the feature and what contact to call

(boot server)

phone1.cfg

when going off hook.

 

 

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

 

 

<autoOffHook/> on page A-112.

 

 

 

Call Hold

The purpose of hold is to pause activity on one call so that the user may use the phone for another task, such as to make or receive another call. Network signaling is employed to request that the remote party stop sending media and to inform them that they are being held. A configurable local hold reminder feature can be used to remind the user that they have placed calls on hold. The call hold reminder is always played through the speakerphone.

4 - 17

Administrator’s Guide SoundPoint IP / SoundStation IP

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

Central

Configuration file:

Specify whether RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or

(boot server)

sip.cfg

a=inactive) outgoing hold signaling is used.

 

 

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

 

 

Specify local hold reminder options.

 

 

For more information, refer to Hold, Local Reminder

 

 

<hold/><localReminder/> on page A-67.

 

 

Specify the Music on Hold URI.

 

 

For more information, refer to Music on Hold <musicOnHold/> on

 

 

page A-17.

 

 

 

 

Configuration file:

Specify the Music on Hold URI.

 

phone1.cfg

For more information, refer to Music on Hold <musicOnHold/> on

 

 

page A-17.

 

 

 

Local

Web Server

Specify whether or not to use RFC 2543 (c=0.0.0.0) outgoing hold

 

(if enabled)

signaling. The alternative is RFC 3264 (a=sendonly or a=inactive).

 

Navigate to: http://<phoneIPAddress>/appConf.htm#ls

 

 

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

 

Local Phone User

Use the SIP Configuration menu to specify whether or not to use RFC

 

Interface

2543 (c=0.0.0.0) outgoing hold signaling. The alternative is RFC 3264

 

 

(a=sendonly or a=inactive).

 

 

 

Call Transfer

Call transfer enables the user (party A) to move an existing call (party B) into a new call between party B and another user (party C) selected by party A. The phone offers three types of transfers:

Blind transfers—The call is transferred immediately to party C after party A has finished dialing party C’s number. Party A does not hear ring-back.

Attended transfers—Party A dials party C’s number and hears ring-back and decides to complete the transfer before party C answers. This option can be disabled.

Consultative transfers—Party A dials party C’s number and talks privately with party C after the call is answered, and then completes the transfer or hangs up.

4 - 18

 

 

Configuring Your System

 

 

 

 

Configuration changes can performed centrally at the boot server:

 

 

 

Central

Configuration file:

Specify whether to allow a transfer during the proceeding state of a

(boot server)

sip.cfg

consultation call.

 

 

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

 

 

Specify whether a transfer is blind or not.

 

 

For more information, refer to Call Handling Configuration <call/>

 

 

on page A-64.

 

 

 

Local / Centralized Conferencing

The phone can conference together the local user with the remote parties of a configurable number of independent calls by using the phone’s local audio processing resources for the audio bridging. There is no dependency on network signaling for local conferences.

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

 

Note

Conferences are not available when the G.729 codec is enabled on the

 

 

 

SoundStation IP 4000 conference phone.

 

 

 

 

 

 

 

 

Configuration changes can performed centrally at the boot server:

 

 

 

 

Central

 

Configuration file:

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

(boot server)

 

sip.cfg

 

is on hold).

 

 

 

 

For more information, refer to Call Handling Configuration <call/>

 

 

 

 

on page A-64.

 

 

 

 

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

 

 

 

 

conference.

 

 

 

 

For more information, refer to Call Handling Configuration <call/>

 

 

 

 

on page A-64.

 

 

 

 

Specify which type of conference to establish and the address of the

 

 

 

 

centralized conference resource.

 

 

 

 

For more information, refer to Conference Setup <conference/>

 

 

 

 

on page A-16.

 

 

 

 

 

 

 

 

Manage Conferences

 

Note

This feature is supported on the SoundPoint IP 550, 560, 650, and 670 desktop

 

 

 

phones and the SoundStation IP 7000 conference phone.

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

4 - 19

Administrator’s Guide SoundPoint IP / SoundStation IP

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 boot server:

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

For more information, refer to Feature <feature/> on page A-92.

 

 

 

Call Forward

The phone provides a flexible call forwarding feature to forward calls to another destination. Call forwarding can be applied in the following cases:

Automatically to all calls

Calls from a specific caller (extension)

When the phone is busy

When Do Not Disturb is active

After an extended period of alerting

The user can elect to manually forward calls while they are in the alerting state to a predefined or manually specified destination. The call forwarding feature works in conjunction with the distinctive incoming call treatment feature (refer to Distinctive Incoming Call Treatment on page 4-6). The user’s ability to originate calls is unaffected by all call forwarding options. Each registration has its own forwarding properties.

Server-based call forwarding is active if the feature is enabled on both the phone and the server and the phone is registered. If server-based call forwarding is enabled on any of the phone’s registrations, the other registrations are not affected.

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

There is no indication on the phone’s user interface whether or not server-based call forwarding is active.

If server-based call forwarding is enabled, but inactive, and the user selects the call forward soft key, the “moving arrow” icon does not appear on the user’s phone (incoming calls are not forwarded).

Note

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

 

Appearance is enabled.

 

 

4 - 20

Configuring Your System

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.) .

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

Central

Configuration file:

Enable or disable server-based call forwarding.

(boot server)

sip.cfg

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

 

 

Enable or disable display of Diversion header and the order in which

 

 

to display the caller ID and number.

 

 

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

 

 

 

 

Configuration file:

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

 

phone1.cfg

feature.

 

 

For more information, refer to Registration <reg/>on page A-107.

 

 

Set all call diversion settings including a global forward-to contact and

 

 

individual settings for call forward all, call forward busy, call forward

 

 

no-answer, and call forward do-not-disturb.

 

 

For more information, refer to Diversion <divert/> on page A-114.

 

 

 

Local

Web Server

Set all call diversion settings.

 

(if enabled)

Navigate to: http://<phoneIPAddress>/reg.htm

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

 

Local Phone User

The user can set the call-forward-all setting from the idle display

 

Interface

(enable/disable and specify the forward-to contact) as well as divert

 

 

callers while the call is alerting.

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

Directed Call Pick-Up

Calls to another phone can be picked up by dialing the extension of the other phone. This feature depends on support from a SIP server.

Configuration changes can performed centrally at the boot server:

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

For more information, refer to Feature <feature/> on page A-92.

 

 

 

4 - 21

Administrator’s Guide SoundPoint IP / SoundStation IP

Group Call Pick-Up

Calls to another phone within a pre-defined group can be picked up without dialing the extension of the other phone. This feature depends on support from a SIP server.

Configuration changes can performed centrally at the boot server:

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

For more information, refer to Feature <feature/> on page A-92.

 

 

 

Call Park/Retrieve

An active call can be parked, and the parked call can be retrieved by another phone. This feature depends on support from a SIP server.

Configuration changes can performed centrally at the boot server:

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

For more information, refer to Feature <feature/> on page A-92.

 

 

 

Last Call Return

The phone allows server-based last call return. This feature depends on support from a SIP server.

Configuration changes can performed centrally at the boot server:

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

For more information, refer to Feature <feature/> on page A-92.

 

 

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

 

 

For more information, refer to Call Handling Configuration <call/>

 

 

 

on page A-64.

 

 

 

 

Setting Up Advanced Features

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

Configurable Feature Keys

Multiple Line Keys per Registration

Multiple Call Appearances

Shared Call Appearances

4 - 22

Configuring Your System

Bridged Line Appearance

Busy Lamp Field

Customizable Fonts and Indicators

Instant Messaging

Multilingual User Interface

Downloadable Fonts

Synthesized Call Progress Tones

Microbrowser

Real-Time Transport Protocol Ports

Network Address Translation

Corporate Directory

Recording and Playback of Audio Calls

Daisy-Chaining Phones

Provisioning Phones Over CLink

Enhanced Feature Keys

Configurable Soft Keys

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

Voice Mail Integration

Multiple Registrations

Automatic Call Distribution

Server Redundancy

Presence

Microsoft Live Communications Server 2005 Integration

Access URL in SIP Message

Static DNS Cache

Display of Warnings from SIP Headers

4 - 23

Administrator’s Guide SoundPoint IP / SoundStation IP

Configurable Feature Keys

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

Note

No feature keys on the SoundStation IP 4000, 6000, or 7000 can be remapped.

 

Since there is no Redial key on the SoundPoint IP 330/320 phone, the redial

 

function cannot be remapped.

 

 

The rules for remapping of key functions are:

The phone keys that have removable key caps can be mapped to the following:

Any function that is implemented as a removable key cap on any of the phones (Directories, Applications, Conference, Transfer, Redial, Menu, Messages, Do Not Disturb, Call Lists)

A speed-dial

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 performed centrally at the boot server:

Central

Configuration File:

Set the key scrolling timeout, key functions, and sub-pointers for each

(boot server)

sip.cfg

key (usually not necessary).

 

 

For more information, refer to Keys <key/> on page A-75.

 

 

 

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

Feature Key Layouts on page C-12.

4 - 24

Configuring Your System

Multiple Line Keys per Registration

More than one Line Key can be allocated to a single registration (phone number or line) on SoundPoint IP phones. The number of Line Keys allocated per registration is configurable.

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

Central

Configuration file:

Specify the number of line keys to assign per registration.

(boot server)

phone1.cfg

For more information, refer to Registration <reg/> on page A-107.

 

 

 

Local

Web Server

Specify the number of line keys to assign per registration.

 

(if enabled)

Navigate to http://<phoneIPAddress>/reg.htm

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

 

Local Phone User

Specify the number of line keys to assign per registration using the

 

Interface

SIP Configuration menu. Either the Web Server or the boot server

 

 

configuration files or the local phone user interface should be used to

 

 

configure registrations, not a mixture of these options. When the SIP

 

 

Configuration menu is used, it is assumed that all registrations use

 

 

the same server.

 

 

 

Multiple Call Appearances

The phone supports multiple concurrent calls. The hold feature can be used to pause activity on one call and switch to another call. The number of concurrent calls per line key is configurable. Each registration can have more than one line key assigned to it (refer to the previous section, Multiple Line Keys per Registration).

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

Central

Configuration file:

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

(boot server)

sip.cfg

line key.

 

 

For more information, refer to Call Handling Configuration <call/>

 

 

on page A-64.

 

 

 

 

Configuration file:

Specify per-registration the number of calls that can be active or on

 

phone1.cfg

hold per line key assigned to that registration. This will override the

 

 

default value specified in sip.cfg.

 

 

For more information, refer to Registration <reg/> on page A-107.

 

 

 

4 - 25

Administrator’s Guide SoundPoint IP / SoundStation IP

Local

Web Server

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

 

(if enabled)

line key and the number of calls per registration that can be active or

 

on hold per line key assigned to that registration.

 

 

 

 

Navigate to http://<phoneIPAddress>/appConf.htm#ls and

 

 

http://<phoneIPAddress>/reg.htm

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

 

Local Phone User

Specify per-registration the number of calls that can be active or on

 

Interface

hold per line key assigned to that registration using the SIP

 

 

Configuration menu. Either the Web Server or the boot server

 

 

configuration files or the local phone user interface should be used to

 

 

configure registrations, not a mixture of these options. When the SIP

 

 

Configuration menu is used, it is assumed that all registrations use

 

 

the same server.

 

 

 

Shared Call Appearances

Calls and lines on multiple phones can be logically related to each other. A call that is active on one phone will be presented visually to phones that share that call appearance. Mutual exclusion features emulate traditional PBX or key system privacy for shared calls. Incoming calls can be presented to multiple phones simultaneously. Users at the different locations have the ability to interrupt remote active calls.

This feature is dependent on support from a SIP server that binds the appearances together logically and looks after the necessary state notifications and performs an access control function. For more information, refer to Shared Call Appearance Signaling on page B-10.

4 - 26

 

 

Configuring Your System

 

 

 

 

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

 

 

 

Central

Configuration file:

Specify whether diversion should be disabled on shared lines.

(boot server)

sip.cfg

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

 

 

A-67.

 

 

Specify line-seize subscription period.

 

 

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

 

 

Specify standard or non-standard behavior for processing line-seize

 

 

subscription for mutual exclusion feature.

 

 

For more information, refer to Special Events <specialEvent/> on

 

 

page A-16.

 

 

 

 

Configuration file:

Specify per-registration line type (private or shared), barge-in

 

phone1.cfg

capabilities, and line-seize subscription period if using per-registration

 

 

servers. A shared line will subscribe to a server providing call state

 

 

information.

 

 

For more information, refer to Registration <reg/> on page A-107.

 

 

Specify per-registration whether diversion should be disabled on

 

 

shared lines.

 

 

For more information, refer to Diversion <divert/> on page A-114.

 

 

 

Local

Web Server

Specify line-seize subscription period.

 

(if enabled)

Navigate to http://<phoneIPAddress>/appConf.htm#se

 

 

Specify standard or non-standard behavior for processing line-seize

 

 

subscription for mutual exclusion feature.

 

 

Navigate to http://<phoneIPAddress>/appConf.htm#ls

 

 

Specify per-registration line type (private or shared) and line-seize

 

 

subscription period if using per-registration servers, and whether

 

 

diversion should be disabled on shared lines.

 

 

Navigate to http://<phoneIPAddress>/reg.htm

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

 

Local Phone User

Specify per-registration line type (private or shared) using the SIP

 

Interface

Configuration menu. Either the Web Server or the boot server

 

 

configuration files or the local phone user interface should be used to

 

 

configure registrations, not a mixture of these options. When the SIP

 

 

Configuration menu is used, it is assumed that all registrations use

 

 

the same server.

 

 

 

Bridged Line Appearance

Calls and lines on multiple phones can be logically related to each other. A call that is active on one phone will be presented visually to phones that share that line. Mutual exclusion features emulate traditional PBX or key system privacy for shared calls. Incoming calls can be presented to multiple phones

4 - 27

Administrator’s Guide SoundPoint IP / SoundStation IP

simultaneously. This feature is dependent on support from a SIP server that binds the appearances together logically and looks after the necessary state notifications and performs an access control function. For more information, refer to Bridged Line Appearance Signaling on page B-10.

 

Note

In the configuration files, bridged lines are configured by “shared line” parameters.

 

 

 

 

 

 

 

 

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

 

 

 

 

Central

 

Configuration file:

Specify whether diversion should be disabled on shared lines.

(boot server)

 

sip.cfg

 

For more information, refer to Call Handling Configuration <call/>

 

 

 

 

on page A-64.

 

 

 

 

 

 

Configuration file:

Specify per-registration line type (private or shared) and the shared

 

 

phone1.cfg

line third party name. A shared line will subscribe to a server

 

 

 

 

providing call state information.

 

 

 

 

For more information, refer to Registration <reg/> on page A-107.

 

 

 

 

Specify per-registration whether diversion should be disabled on

 

 

 

 

shared lines.

 

 

 

 

For more information, refer to Diversion <divert/> on page A-114.

 

 

 

 

Local

 

Web Server

Specify per-registration line type (private or shared) and third party

 

 

(if enabled)

name, and whether diversion should be disabled on shared lines.

 

 

Navigate to http://<phoneIPAddress>/reg.htm

 

 

 

 

 

 

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

 

 

override global settings unless deleted through the Reset Local

 

 

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

 

 

removed from the boot server.

 

 

 

 

 

 

Local Phone User

Specify per-registration line type (private or shared) and the shared

 

 

Interface

 

line third party name using the SIP Configuration menu. Either the

 

 

 

 

Web Server or the boot server configuration files or the local phone

 

 

 

 

user interface should be used to configure registrations, not a mixture

 

 

 

 

of these options. When the SIP Configuration menu is used, it is

 

 

 

 

assumed that all registrations use the same server.

 

 

 

 

 

Busy Lamp Field

Note

This feature is available only on SoundPoint IP 320/330, 430, 550, 560, 600, 601,

 

650, and 670 phones. However, on the SoundPoint IP 320/330, the LED is not lit.

 

Depending on your call server, certain aspects of this feature work may not work as

 

described below.

 

 

4 - 28

Configuring Your System

The Busy Lamp Field (BLF) feature enhances support for a phone-based attendant console. It allows monitoring the hook status and remote party information of users through the busy lamp fields and displays on an attendant console phone.

In the SIP 3.1 release, the BLF feature is updated for the following:

Visual indication when a remote line is in an alerting state

Display of the caller ID of calls on remotely monitored lines

Single button “Directed Call Pickup” on a remote line

 

 

 

For more information, refer to “Quick Tip 37381: Enhanced BLF“ at

 

 

 

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

 

 

 

echnical_Bulletins_pub.html .

 

 

 

 

 

 

 

 

Polycom recommends that the BLF not be used in conjunction with the Microsoft

 

 

 

Live Communications Server 2005 feature. For more information, refer to Microsoft

 

 

 

Live Communications Server 2005 Integration on page 4-61.

 

Note

 

 

 

 

 

 

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

 

 

 

A-7). You can also use UDP transport on SoundPoint IP 650 and 670 phones.

 

 

 

 

 

 

 

 

Configuration changes can performed centrally at the boot server:

 

 

 

 

Central

 

Configuration file:

Specify the list SIP URI and index of the registration which will be

(boot server)

 

phone1.cfg

used to send a SUBSCRIBE to the list SIP URI specified in

 

 

 

 

attendant.uri.

 

 

 

 

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

 

 

 

 

A-121.

 

 

 

 

 

Customizable Fonts and Indicators

The phone’s user interface can be customized by changing the fonts and graphic icons used on the display and the LED indicator patterns. Pre-existing fonts embedded in the software can be overwritten or new fonts can be downloaded. The bitmaps and bitmap animations used for graphic icons on the display can be changed and repositioned. LED flashing sequences and colors can be changed.

4 - 29

Administrator’s Guide SoundPoint IP / SoundStation IP

Configuration changes can performed centrally at the boot server:

Central (boot

Configuration File:

Specify fonts to overwrite existing ones or specify new fonts.

server)

sip.cfg

For more information, refer to Fonts <font/> on page A-72.

 

 

Specify which bitmaps to use.

 

 

For more information, refer to Bitmaps <bitmap/>on page A-80.

 

 

Specify how to create animations and LED indicator patterns.

 

 

For more information, refer to Indicators <ind/> on page A-80.

 

 

 

Instant Messaging

The phone supports sending and receiving instant text messages. The user is alerted to incoming messages visually and audibly. The user can view the messages immediately or when it is convenient. For sending messages, the user can either select a message from a preset list of short messages or an alphanumeric text entry mode allows the typing of custom messages using the dial pad. Message sending can be initiated by replying to an incoming 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.

Configuration changes can be performed centrally at the boot server:

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

For more information, refer to Feature <feature/> on page A-92.

 

 

 

Multilingual User Interface

Note This feature is not available on SoundPoint IP 301 phones.

The system administrator or the user can select the language. Support for major western European languages is included and additional languages can be easily added. Support for Asian languages (Chinese, Japanese, and Korean) is also included, but will display only on the SoundPoint IP 550, 560, 650, and 670 and SoundStation IP 4000, 6000, and 7000’s higher resolution displays. A WGL4 character set is displayed 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 550, 560, 600, 601, and 650 and 670 and SoundStation IP platforms), refer to Multilingual <ml/> on page A-22. (Note that within a Unicode range, some characters may not be supported due to their infrequent usage.)

4 - 30

Configuring Your System

 

 

 

The SoundPoint IP and SoundStation IP user interface is available in the

 

 

 

following languages by default: Chinese (if displayable), Danish, Dutch,

 

 

 

English, French, German, Italian, Japanese (if displayable), Korean (if

 

 

 

displayable), Norwegian, Polish, Portuguese, Russian, Slovenian, Spanish,

 

 

 

and Swedish.

 

 

Note

 

 

 

Slovenian is not supported on the SoundStation IP 4000.

 

Note

 

 

 

 

 

 

The multilingual feature relies on dictionary files resident on the boot server. The

 

 

 

dictionary files are downloaded from the boot server whenever the language is

 

 

 

changed or at boot time when a language other than the internal US English

 

 

 

language has been configured. If the dictionary files are inaccessible, the language

 

 

 

will revert to the internal language.

 

Note

 

 

 

 

 

 

Currently, the multilingual feature is only available in the application. At this time,

 

 

 

the bootROM application is available in English only.

 

 

 

 

 

 

 

 

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

 

 

 

 

 

Central

 

Configuration file:

 

Specify the boot-up language and the selection of language choices

(boot server)

 

sip.cfg

 

 

to be made available to the user.

 

 

 

 

 

For more information, refer to Multilingual <ml/> on page A-22.

 

 

 

 

 

For instructions on adding new languages, refer to To add new

 

 

 

 

 

languages to those included with the distribution: on page A-23.

 

 

 

 

 

Local

 

Local Phone User

 

The user can select the preferred language under the Settings menu.

 

 

Interface

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

 

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

 

 

 

override global settings unless deleted through the Reset Local

 

 

 

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

 

 

 

removed from the boot server.

 

 

 

 

 

Downloadable Fonts

 

 

 

 

 

 

New fonts can be loaded onto the phone. For guidelines on downloading

 

 

 

fonts, refer to Fonts <font/> on page A-72.

 

Note

 

 

Downloadable fonts are not supported on the SoundStation IP 6000 and 7000.

 

 

 

 

 

 

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Administrator’s Guide SoundPoint IP / SoundStation IP

Synthesized Call Progress Tones

In order to emulate the familiar and efficient audible call progress feedback generated by the PSTN and traditional PBX equipment, call progress tones are synthesized during the life cycle of a call. These call progress tones are easily configurable for compatibility with worldwide telephony standards or local preferences.

Configuration changes can performed centrally at the boot server:

Central

Configuration file:

Specify the basic tone frequencies, levels, and basic repetitive

(boot server)

sip.cfg

cadences.

 

 

For more information, refer to Chord-Sets <chord/> on page A-29.

 

 

Specify downloaded sampled audio files for advanced call progress

 

 

tones.

 

 

For more information, refer to Sampled Audio for Sound Effects

 

 

<saf/> on page A-30.

 

 

Specify patterns.

 

 

For more information, refer to Patterns <pat/> on page A-32 and

 

 

Call Progress Patterns on page A-33.

 

 

 

Microbrowser

The SoundPoint IP 430, 501, 550, 560, 600, 601, 650, and 670 phones and the SoundStation IP 4000, 6000, and 7000 phones supports an XHTML Microbrowser. This can be launched by pressing the Applications key or it can be accessed through the Menu key by selecting Features, and then

Applications.

Note

As of SIP 2.2.0, the Services key and menu entry are renamed Applications,

 

however the functionality remains the same.

 

 

Two instances of the Microbrowser may run concurrently:

An instance with standard interactive user interface

An instance that does not support user input, but appears in a window on the idle display

For more information, refer to the Web Application Developer’s Guide.

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Configuring Your System

 

 

 

 

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

 

 

 

Central

Configuration file:

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

(boot server)

sip.cfg

limits.

 

 

For more information, refer to Microbrowser <mb/> on page A-95.

 

 

Specify the telephone notification and state polling events to be

 

 

recorded and location of the push server.

 

 

For more information, refer to Applications <apps/> on page A-98.

 

 

 

Local

Web Server

Specify the Applications browser home page and proxy to use.

 

(if enabled)

Navigate to http://<phoneIPAddress>/coreConf.htm#mb

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

Real-Time Transport Protocol Ports

The phone is compatible with RFC 1889 - RTP: A Transport Protocol for Real-Time Applications - and the updated RFCs 3550 and 3551. Consistent with RFC 1889, the phone treats all RTP streams as bi-directional from a control perspective and expects that both RTP end points will negotiate the respective destination IP addresses and ports. This allows real-time transport control protocol (RTCP) to operate correctly even with RTP media flowing in only a single direction, or not at all. It also allows greater security: packets from unauthorized sources can be rejected.

The phone can filter incoming RTP packets arriving on a particular port by IP address. Packets arriving from a non-negotiated IP address can be discarded.

The phone can also enforce symmetric port operation for RTP packets: packets arriving with the source port set to other than the negotiated remote sink port can be rejected.

The phone can also jam the destination transport port to a specified value regardless of the negotiated port. This can be useful for punching through firewalls. When this is enabled, all RTP traffic will be sent to the specified port and will be expected to arrive on that port as well. Incoming packets are sorted by the source IP address and port, allowing multiple RTP streams to be multiplexed.

The RTP port range used by the phone can be specified. Since conferencing and multiple RTP streams are supported, several ports can be used concurrently. Consistent with RFC 1889, the next higher odd port is used to send and receive RTCP.

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Administrator’s Guide SoundPoint IP / SoundStation IP

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

Central

Configuration file:

Specify whether to filter incoming RTP packets by IP address,

(boot server)

sip.cfg

whether to require symmetric port usage, whether to jam the

 

 

destination port and specify the local RTP port range start.

 

 

For more information, refer to RTP <rtp/> on page A-57.

 

 

 

Local

Web Server

Specify whether to filter incoming RTP packets by IP address,

 

(if enabled)

whether to require symmetric port usage, whether to jam the

 

destination port and specify the local RTP port range start.

 

 

 

 

Navigate to: http://<phoneIPAddress>/netConf.htm#rt

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

Network Address Translation

The phone can work with certain types of network address translation (NAT). The phone’s signaling and RTP traffic use symmetric ports (the source port in transmitted packets is the same as the associated listening port used to receive packets) and the external IP address and ports used by the NAT on the phone’s behalf can be configured on a per-phone basis.

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

Central

Configuration file:

Specify the external NAT IP address and the ports to be used for

(boot server)

sip.cfg

signaling and RTP traffic.

 

 

For more information, refer to Network Address Translation

 

 

<nat/> on page A-120.

 

 

 

Local

Web Server

Specify the external NAT IP address and the ports to be used for

 

(if enabled)

signaling and the RTP traffic.

 

Navigate to: http://<phoneIPAddress>/netConf.htm#na

 

 

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

Corporate Directory

Note

This feature requires a license key for activation. Using this feature may require

 

purchase of a license key or activation by Polycom channels. For more information,

 

contact your Certified Polycom Reseller.

 

 

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Configuring Your System

The SoundPoint IP phones can be configured to interface with a corporate directory server that supports the Lightweight Directory Access Protocol (LDAP) version 3. (Microsoft’s Active Directory is included.) Both corporate directories that support server-side sorting and those that do not are supported. In the latter case, the sorting is performed on the phone.

Polycom recommends using corporate directories that have server-side sorting.

Polycom recommends that you consult your LDAP Administrator when making any configuration changes for this feature.

The corporate directory can be browsed or searched. Entries retrieved from the LDAP server can be saved to the local contact directory on the phone. Phone calls can be placed based on the phone number contained in the LDAP entry.

The corporate directory interface shall be read only, so that editing or deleting existing directory entries as well as adding new directory entries from the phone shall not be possible.

All attributes are considered to be Unicode text. Validity checking will be performed when a call is placed or the entry is saved to the local contact directory.

The corporate directory LDAP server status can be reviewed through the Status menu (Status > CD Server Status).

For more information, refer to “Technical Bulletin 41137: Best Practices When Using Corporate Directory on SoundPoint IP and SoundStation IP Phones“ at http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_T echnical_Bulletins_pub.html .

Configuration changes can performed centrally at the boot server:

Central

Configuration file:

Specify the location of the corporate directory’s LDAP server, the

(boot server)

sip.cfg

LDAP attributes, how often to refresh the local cache from the LDAP

 

 

server, and other miscellaneous parameters.

 

 

For more information, refer to Corporate Directory <corp/> on

 

 

page A-69.

 

 

 

Local

Local Phone User

Enable or disable persistent viewing through the Settings menu

 

Interface

(Settings > Basic > Preferences > Corporate Directory > View

 

 

Persistency).

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfg on the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfg is

 

 

removed from the boot server.

 

 

 

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Administrator’s Guide SoundPoint IP / SoundStation IP

This section contains the following information:

Corporate Directory LDAP Attributes

Browsing the Corporate Directory

Configuration File Example

Corporate Directory LDAP Attributes

The entry attributes in the corporate directory are mapped through sip.cfg configuration file attributes to the LDAP attributes first_name, last_name, phone_number, and others so the SIP application knows how to use them for searching, dialing, or saving to the local contact directory. Multiple attributes of the same type are allowed.

Note

The maximum of eight attributes can be configured in sip.cfg .

 

 

The configuration order dictates how the attributes are displayed and sorted. The first attribute is the primary sort index and the second attribute is the secondary sort index. The other attributes are not used in sorting.

To limit the amount of data displayed in the corporate directory, filtering of the entries can be configured for all attribute types. Filtering can be configured to be retained if the phone reboots.

For more information on LDAP attributes, refer to RFC 4510 - Lightweight Directory Access Protocol (LDAP): Technical Specification Road Map.

Browsing the Corporate Directory

The SoundPoint IP or SoundStation IP phone will establish a session with the corporate directory and download enough entries to fill its cache:

when the corporate directory is first accessed

when the phone boots up if the background synchronization parameter is enabled

The requested entries are based on the configured attributes (see previous section).

If the background synchronization parameter is enabled, a timer is initiated to permit a periodic download from the corporate directory.

Entries are sorted according to the order in which the first two attributes are configured (for example, last name, then first name).

The browse position within the corporate directory as well as the attribute filters are maintained for subsequent corporate directory access can be saved (if so configured).

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Configuring Your System

Configuration File Example

The following excerpt from the sip.cfg configuration file shows an example where downloaded entries are limited to any where the phone number is in the 604 area code:

dir.corp.address=”

dir.corp.port=”389”

dir.corp.transport=”TCP”

dir.corp.baseDN=”cn=Users,dc=yourcompany,dc=local

dir.corp.user=”ldapadmin”

dir.corp.password=”12345678”

dir.corp.filterPrefix=”(objectclass=person)”

dir.corp.scope=”sub”

dir.corp.attribute.1.name=”sn” dir.corp.attribute.1.label=”Last Name” dir.corp.attribute.1.type=”last_name” dir.corp.attribute.1.filter=”” dir.corp.attribute.1.sticky=”0” dir.corp.attribute.2.name=”givenName” dir.corp.attribute.2.label=”First Name” dir.corp.attribute.2.type=”first_name” dir.corp.attribute.2.filter=”” dir.corp.attribute.2.sticky=”0” dir.corp.attribute.3.name=”telephoneNumber” dir.corp.attribute.3.label=”Phone Number” dir.corp.attribute.3.type=”phone_number” dir.corp.attribute.3.filter=”604” dir.corp.attribute.3.sticky=”0” dir.corp.backGroundSync=”0” dir.corp.backGroundSync.period=”86400” dir.corp.viewPersistence=”1”

Recording and Playback of Audio Calls

Note

This feature requires a license key for activation. Using this feature may require

 

purchase of a license key or activation by Polycom channels. For more information,

 

contact your Certified Polycom Reseller.

 

 

 

SoundPoint IP phones that have a USB port can be configured to allow

 

recording of audio calls on a supported USB device. Only the SoundPoint IP

 

650 and 670 have a functioning USB port.

 

The filenames of the recorded .wav files will include a date/time stamp (for

 

example, 20Apr2007_190012.wav was created on April 20, 2007 at 19:00:12).

 

An indication of the recording time remaining—the space available of the

 

attached USB storage media—appears on the graphic display. The user can

 

browse through all recorded files through the menu shown on the graphic

 

display.

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Administrator’s Guide SoundPoint IP / SoundStation IP

Note

Notify your users that they may be required by federal, state, and/or local laws to

 

notify some or all called parties when they are recording.

 

 

Playback of recorded files can occur on the phone as well as on other devices, such as a Windows® or Apple® based computer using an application like Windows Media Player® or iTunes®.

The user controls which calls are recorded and played back.

For a list of supported USB devices, refer to “Technical Bulletin 38084: SoundPoint IP 650 and 670 Supported USB Devices for Recording“ at http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_T echnical_Bulletins_pub.html .

Configuration changes can be performed centrally at the boot server:

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

For more information, refer to Feature <feature/> on page A-92.

 

 

 

Daisy-Chaining Phones

You can join two SoundStation IP family conference phones together through the use of a CLink cable and the Multi-Interface Module. The graphic display of each phone shows the same user interface and phone numbers. The SoundStation IP family phone that has the Ethernet connection is referred to as the primary. The SoundStation IP family phone that does not have the Ethernet connection is referred to as the secondary. The primary/secondary relationship of the phones is determined by their MAC address, registration status, and the configuration files.

Power Adapter

Multi-Interface

Module

5

12-foot Ethernet Cable

Interconnect Cable

25-foot

 

Network Cable

4

4 - 38

Configuring Your System

Instructions for daisy-chaining SoundStation IP family conference phones are available in the SoundStation IP 7000 User Guide.

Provisioning Phones Over CLink

Normally the SoundStation IP family conference phone is provisioned over the Ethernet by the boot server. However, when two SoundStation IP family phones are daisy-chained together, the one that is not directly connected to the Ethernet can still be provisioned (known as the secondary). The provisioning over CLink feature is automatically enabled when a SoundStation IP family phone is not connected to the Ethernet. Both SoundStation IP family phones must be running the same version of the SIP application.

The steps for provisioning the secondary SoundStation IP family phone are the same as for the primary SoundStation IP family phone. You can reboot the primary without rebooting the secondary. However, the primary and secondary should be rebooted together for the primary/secondary relationship to be recognized. If you power up both SoundStation IP family phones, the primary will power up first.

Currently, provisioning over CLink is supported for the following configurations of SoundStation IP family conference phones:

Two SoundStation IP family conference phone daisy-chained together

Two SoundStation IP family conference phone daisy-chained together with one external microphone, specifically designed for the SoundStation IP family conference phone

Refer to Daisy-Chaining Phones on page 4-38 for an illustration of two SoundStation IP family conference phone daisy-chained together.

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

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

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

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

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Administrator’s Guide SoundPoint IP / SoundStation IP

Enhanced Feature Keys

Note

The Enhanced Feature Key feature from SIP 3.0 is compatible with Enhanced

 

Feature Key feature from SIP 3.1 . However, improvements have been made, and

 

Polycom recommends that existing configuration files be reviewed and updated.

 

 

Customers replacing legacy telephony PBX or key system would like to get equivalent functionality from their new VoIP telephony system. With SIP 3.0, this feature allowed system administrators to program the speed-dials on their phones to interact with the phone user to implement commonly used functions such as “Call Park” in an intuitive fashion.

This capability applies to the SoundPoint IP 301, 320, 330, 430, 501, 550, 560, 601, 650, and 670phones. The enhanced feature key functionality is implemented using Star Code sequences and SIP messaging.

The enhanced feature key definition language was defined to follow current configuration file standards and to be extensible.

The particular Star Code sequence and the associated prompts displayed on the SoundPoint IP phone for the enhanced feature are defined by macros. These macros are case sensitive.

An enhanced feature key can be accessed from all instances where the speed-dial is accessible, for example, unused line keys, speed-dial lists or programmed to “hard” function keys.

This section provides detailed information on:

Enhanced Feature Key Definition Language

Macro Definition

Configuration File Changes

Useful Tips

Examples

Enhanced Feature Key Definition Language

This section defines the additional fields to be entered into a configuration file for controlling the enhanced feature key behavior. The definition language follows the XML style notation. The following elements are part of the definition language:

<efk/>

<efklist/>

<efkprompt/>

<version/>

Special Characters

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Configuring Your System

<efk/>

This element indicates the start of enhanced feature key definition section. The efk element has the following format:

<efk> ... </efk>

<efklist/>

This element describes behavior of enhanced feature key.

The different blocks of the enhanced feature key definitions are uniquely identified by number following efk.efklist prefix (for example, efk.efklist.1.<suffix>).

Note

 

In SIP 3.1, a maximum of 50 element groups is supported, however, the exact

 

 

number is dependent on available RAM and processing speed. The disabled

 

 

elements are included in the total count.

 

 

 

 

 

 

This element contains the following parameters:

 

 

 

 

Name

Interpretation

 

 

 

 

mname

This is the unique identifier that is used for the

 

 

 

speed-dial configuration to reference the enhanced

 

 

 

feature key entry. It cannot start with a digit.

 

 

 

This parameter must have a value and it cannot be Null.

 

 

 

 

status

This parameter has the following values:

 

 

 

If set to 1, this key is enabled.

 

 

 

If set to 0 or Null, this key is disabled.

 

 

 

If this parameter is omitted, the value 0 is used.

 

 

 

 

label

This field defines the text string that will be used as a

 

 

 

label on any user text entry screens during enhanced

 

 

 

feature key operation. The value can be any string

 

 

 

including the null string (in this case, no label appears).

 

 

 

If this parameter is omitted, the Null string is used.

 

 

 

Note: If you exceed the phone physical layout text

 

 

 

limits, the text will be shortened and "..." will be

 

 

 

appended.

 

 

 

 

 

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Administrator’s Guide SoundPoint IP / SoundStation IP

Name

Interpretation

 

 

type

The SIP method to be performed once the macro starts

 

executing. This parameter has the following values:

 

If set to “invite “, the action required is performed

 

using the SIP INVITE method.

 

Note: This parameter is included for backwards

 

compatability only. Do not use if at all possible. If the

 

action.string contains types, this parameter is ignored. If

 

this parameter is omitted, the default is INVITE.

 

 

action.string

The action string contains a macro definition of the

 

action to be performed.

 

For more information, refer to Macro Definition on page

 

4-44.

 

This parameter must have a value and it cannot be Null.

 

 

<efkprompt/>

This element describes the behavior of the user prompts.

The different blocks are uniquely identified by number following efk.efkprompt prefix (for example, efk.efkprompt.1.<suffix>).

Note

 

In SIP 3.0, a maximum of four user prompts were supported. In SIP 3.1, a

 

 

maximum of ten user prompts are supported.

 

 

 

 

 

 

This element contains the following parameters:

 

 

 

 

Name

Interpretation

 

 

 

 

status

This parameter has the following values:

 

 

 

If set to 1, this key is enabled.

 

 

 

If set to 0, this key is disabled.

 

 

 

This parameter must have a value and it cannot be Null.

 

 

 

Note: If a macro attempts to use a prompt that is

 

 

 

disabled or invalid, the macro execution fails.

 

 

 

 

label

This parameter sets the prompt text that will be

 

 

 

presented to the user on the user prompt screen. The

 

 

 

value can be any string including the null string (in this

 

 

 

case, no label appears).

 

 

 

If this parameter is omitted, the Null string is used.

 

 

 

Note: If you exceed the phone physical layout text

 

 

 

limits, the text will be shortened and "..." will be

 

 

 

appended.

 

 

 

 

4 - 42

 

 

 

Configuring Your System

 

 

 

 

 

 

 

Name

Interpretation

 

 

 

 

userfeedback

This parameter specifies the user input feedback

 

 

method. It has the following values:

 

 

If set to “visible”, the text appears as clear text.

 

 

If set to “masked”, the text appears as “*”

 

 

 

characters. For example, if a password is entered.

 

 

If this parameter is omitted, the value “visible” is used.

 

 

If this parameter has an invalid value (including Null),

 

 

this prompt is invalid and all parameters depending on

 

 

this prompt are invalid.

 

 

 

 

type

The type of characters entered by the user. This

 

 

parameter has the following values:

 

 

If set to “numeric “, the characters are interpreted as

 

 

 

numbers.

 

 

If set to “text”, the characters are interpreted as

 

 

 

letters.

 

 

If this parameter is omitted, the value “numeric” is used.

 

 

If this parameter has an invalid value (including Null),

 

 

this prompt is invalid and all parameters depending on

 

 

this prompt are invalid.

 

 

Note: A mix of numeric and text is not supported.

 

 

 

 

<version/>

This element contains the version of the enhanced feature key elements. The version element has the following format:

<version efk.version=”2”/>

If this parameter is omitted or has an invalid value (including Null), the enhanced feature key is disabled.

Note

In SIP 3.0, “1” is the only supported version. In SIP 3.1, “2” is the only supported

 

version.

 

 

Special Characters

The following special characters are used to implement the enhanced feature key functionality:

! — The characters following it are a macro name.

$ — This character delimits the parts of the macro string. This character must exist in pairs, where the delimits the characters to be expanded.

^ — This character indicates that the following characters represent the expanded macro (as in the action string).

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Macro names and action strings cannot contain these characters. If they do, unpredictable results may occur.

Macro Definition

The action.string in the efklist element can be defined by either:

Macro Action

Prompt Macro Substitution

Expanded Macros

Macro Action

The action string is executed in the order it appears. User input is collected before any action is taken.

The action string contains the following fields:

Name

Interpretation

 

 

$L<label>$

This is the label for the entire operation. The value can

 

be any string including the null string (in this case, no

 

label appears). This label will be used if no other

 

operation label collection method worked (up to the

 

point where this field is introduced). Make this the first

 

entry in action string to be sure this label is used;

 

otherwise another label may be used and this one

 

ignored.

 

 

digits

The digits to be sent.

 

The appearance of this this parameter depends on the

 

action string.

 

 

$C<command>$

This is the command. It can appear anywhere in the

 

action string.

 

Supported commands (or shortcuts) include:

 

hangup (hu)

 

hold (h)

 

waitconnect (wc)

 

pause <number of seconds> (p <num sec>) where

 

 

the maximum value is 10

 

 

 

4 - 44

 

 

 

Configuring Your System

 

 

 

 

 

 

 

Name

Interpretation

 

 

 

 

$T<type>$

The embedded action type. Multiple actions can be

 

 

defined.

 

 

Supported action types include:

 

 

invite

 

 

dtmf

 

 

refer

 

 

Note: Polycom recommends that you always define this

 

 

field. If it is not defined, the supplied digits will be dialed

 

 

using INVITE (if no active call) or DTMF (if an active

 

 

call). The use of refer method is call server

 

 

dependentand may require the addition of star codes.

 

 

 

 

$M<macro>$

The embedded macro. The <macro> string must begin

 

 

with a letter.

 

 

If the macro name is not defined, the execution of the

 

 

action string fails.

 

 

 

 

$P<prompt num>N<num

The user input prompt string.

 

digits>$

Refer to Prompt Macro Substitution on this page.

 

 

 

 

 

 

$S<speed dial index>$

The speed dial index. Only digits are valid.

 

 

The action is found in the contact field of the local

 

 

directory entry pointed to by the index.

 

 

 

 

$F<internal function>$

An internal function.

 

 

For more information, refer to Internal Key Functions on

 

 

page C-19.

 

 

 

 

URL

A URL. Only one per action string is supported.

 

 

 

 

Prompt Macro Substitution

The action.string in the efklist element can be defined by a macro substitution string, “PnNn” where:

Pn is the prompt x as defined in the efk.efkprompt.x

Nn is the number of digits or letters that the user can enter. The maximum number is 32. The user needs to press the Enter soft key to complete data entry.

Note

If the maximum number of characters is greater than 32 or less than one, macro

 

execution fails.

 

 

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Administrator’s Guide SoundPoint IP / SoundStation IP

 

 

 

The macros provide a generic and easy to manage way to define the prompt to

 

 

 

be displayed to the user, the maximum number of characters that the user can

 

 

 

input, and action that the phone performs once all user input has been

 

 

 

collected. The macros are case sensitive.

 

 

 

If a macro attempts to use a prompt that is disabled, the macro execution fails.

 

 

 

A prompt is not required for every macro.

 

 

 

Expanded Macros

 

 

 

Expanded macros are prefixed with the “^” character and are inserted directly

 

 

 

into the local directory contact field. For more information, refer to Local

 

 

 

Contact Directory File Format on page 4-10.

 

 

 

Configuration File Changes

 

Note

 

 

 

The configuration file changes and the enhanced feature key definitions can be

 

 

 

included together in one configuration file.

 

 

 

A sample configuration for this feature—including the enhanced feature keys

 

 

 

definitions shown in the following section, Examples— may be included with the

 

 

 

SIP 3.1 release.

 

 

 

Create a new configuration file in the style of sip.cfg in order to make configuration

 

 

 

changes. For more information on why to create another configuration file, refer to

 

 

 

the “Configuration File Management on SoundPoint IP Phones” whitepaper at

 

 

 

www.polycom.com/support/voice/ .

 

 

 

 

 

 

 

 

Configuration changes can performed centrally at the boot server:

 

 

 

 

Central

 

Configuration file:

Turn this feature on or off.

(boot server)

 

sip.cfg

 

For more information, refer to Feature <feature/> on page A-92.

 

 

 

 

 

 

Configuration file:

Specify two calls per line key.

 

 

phone1.cfg

For more information, refer to Registration <reg/> on page A-107.

 

 

 

 

 

 

XML file: <Ethernet

This file holds the macro names which correspond to the mname fields

 

 

address>-directory.

in the configuration file where the enhanced feature keys are defined.

 

 

xml

 

Macro names must be embedded into the contact (cn) fields with the

 

 

 

 

“!” prefix. You can also add labels in the first name (fn) fields.

 

 

 

 

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

 

 

 

 

Format on page 4-10.

 

 

 

 

 

Useful Tips

The following information should be noted:

Activation of the enhanced feature key will fail if configured values are invalid except where noted in previous sections.

All failures are logged at level 4 (minor).

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