LINKSYS 900 Administrator Guide

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Linksys 900 Series IP Phone Administrator Guide

Document Version 3.0

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Linksys 900 Series IP Phone Administrator Guide

Copyright ©2007 Cisco Systems, Inc. All rights reserved.Speci fications are subject to change without notice. Linksys is a registered trademark or trademar k of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries. Other brands and product names are trademarks or registered trademarks of their respective holders.

Compliance and Safety Information This equipment has been tested and found to comply with the limits for a Class B digital de vice in accordance with the sp ecifications in p art 15 of the FCC rules. This product

bears the CE Marking indicating compliance with the 89/336/EEC directive. Standards to which conformity is Declared: EN 61000-4-2:1995, EN 61000-4-3:1997, EN 61000-4-4:1995, EN 61000-4-5:1995, EN 61000-4-6:1996, EN 61000-4-8:1994, EN 61000-4-11:1994, EN 61000-3-2:2001, EN 61000-3-3:1995 & EN 55022:1998

Class B Modifications to this product not authorized by Linksys could void FCC approval, thereby terminating end user authority to use this product. For indoor use only. Read installation instructions before connecting to a po wer source. The electric pl ug and so cket must be accessib le at all times as this is the main method to disc onnect power from the device.

Shock Hazard: Do not operate near water or similar fluid. Do not work with this device during periods of lightning activity. Do not touch wires at the end of cables or within sockets.

One Year Limited Hardware Warranty Linksys provides a one (1) year limited hardware warranty. Linksys warrants to customer that this product conforms to its published specifications and will be free from

defects in material and workmanship at the time of delivery and for a period of one year thereafter. Without limiting the foregoing, this warranty does not cover any defect resulting from (i) any design or specificatio n supplied by an ent ity other t han Links ys, (ii) non-observ ance of techn ical opera ti ng parameters (e.g., exceeding limiti ng values), or (iii) misuse, abuse, abnormal conditions or alteration by anyone other than Linksys. Replacement, Repair, Refund: After the receipt of an RMA (Return Materials Authorization) request, Linksys will attempt to refund, repair or replace this device. To receive an RMA number for this device, contact the party from whom it.

Preface xi

Document Audience xi Linksys 900 Series IP Telephones xi How This Document is Organized xii Document Conventions xii Related Documentation xiii Technical Support xiii

CHAPTER

1 Introducing Linksys 900 Series IP Phones 1-1

Overview 1-1 SPA900 Series Features 1-2

SPA901 Features 1-4 SPA92x, SPA94x, and SPA962 Features 1-4 Ensuring Voice Quality 1-4

Feature Descriptions 1-6

SIP Proxy Redundancy 1-6 Supported Codecs 1-6 Other Features 1-7

Technology Background 1-10

Session Initiation Protocol 1-10 Using 900 Series Phones with a Firewall or Router 1-11 Network Address Translation 1-11

NAT Overview 1-12 NAT Types 1-12 Simple Traversal of UDP Through NAT 1-13 SIP-NAT Interoperation 1-13

CHAPTER

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2 Getting Started 2-1

Where to Go From Here 1-14

Linksys 900 Series IP Phones 2-1

Caring for Your Hardware 2-2 SPA901 2-2

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Contents

Front Panel and Side of Phone 2-2

Back Panel 2-3 SPA92x, SPA94x, and SPA962 Hardware Features 2-3 SPA921 2-4

Front Panel 2-4

Back Panel 2-5 SPA922 2-5 SPA941 2-5

Front Panel 2-6

Back Panel 2-6 SPA942 2-7 SPA962 2-7

Front Panel 2-8

Back Panel 2-8

Establishing Connectivity 2-8

Bandwidth Requirements 2-8 Installing the SPA900 Series IP Phone 2-9 Assembling the Phone and Connecting to the Network 2-9 Attaching the Desk Stand 2-10 Mounting the Phone to the Wall 2-10 Turning on the Phone 2-11

CHAPTER

Using the Administration Web Server 2-11

Connecting to the Administration Web Server 2-11 Administrator Account Privileges 2-12

Web Interface URLs 2-13

Upgrade URL 2-13 Resync URL 2-13 Reboot URL 2-14

Provisioning 2-14

Provisioning Capabilities 2-14

Configuration Profile 2-14

Using the Interactive Voice Response Interface 2-15

Using the IVR Menu 2-15 IVR Options 2-16 Entering a Password through the IVR 2-18

3 Managing Linksys 900 Series IP Phones 3-1

Using the LCD Display 3-1

LCD Display Controls 3-1

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Using Soft Keys 3-3

Entering and Saving Settings 3-4 Localization 3-5 Changing the Display Background (SPA962) 3-7 Call Appearances and Extensions 3-8 Line Key LEDs 3-9 Using Call Features 3-10

Selecting the Audio I/O Device and Line 3-11

Making Calls 3-11

Answering and Ending Calls 3-12

Hold and Resume 3-12

Call Waiting 3-12

Speed Dialing 3-13

Three-Way Conferencing 3-13

Attended Call Transfer 3-13

Blind Call Transfer 3-14

Call Back 3-14

Message Waiting Indication (MWI) 3-14

Accessing Voicemail 3-15

Muting Calls 3-15

Shared Call Appearances 3-15

Personal Directory 3-15

Caller and Called Name Matching 3-16

Dialing Assistance 3-16

Supplementary Services 3-16

Call Logs 3-17

Audio Volume Adjustment 3-18

Managing Ring Tones 3-19

Contents

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Configuring a Dial Plan 3-20

Dial Plan Digit Sequences 3-20

Dial Plan Rules 3-21

Digit Sequence Syntax 3-21 Element Repetition 3-21 Sub-sequence Substitution 3-21 Intersequence Tones 3-22 Number Barring 3-22 Interdigit Timer Master Override 3-22 Local Timer Overrides 3-22 Pause 3-22

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Dial Plan Examples 3-23 Dial Plan Timers 3-23

Interdigit Long Timer 3-24 Interdigit Short Timer 3-24 Dial Plans 3-24

System Administration 3-24

Reboot and Restart 3-25 Factory Reset 3-25 Password Protection 3-25 Managing the Time/Date 3-25 Daylight Saving Time 3-25 Using Star Codes to Activate/Deactivate Services 3-26 Disabling Services 3-28 Error and Log Reporting 3-29

Troubleshooting FAQ 3-29

CHAPTER

4 LCD Command Reference Guide 4-1

1 Directory 4-2

Entering Names and Numbers into the Directory 4-2

Entering Directory Names, Numbers and Ring Default 4-2 2 Speed Dial 4-3 3 Call History 4-3

Redial List 4-3

Answered Calls 4-4

Missed Calls 4-4 4 Ring Tone 4-4 5 Preferences 4-4

5.1 Block Caller ID 4-5

5.2 Block Anonymous Call 4-5

5.3 Do Not Disturb 4-5

5.4 Secure Call 4-5

5.5 Dial Assistance 4-6

5.6 Preferred Audio Device 4-6

6 Call Forward 4-6

6.1 CFWD All Number 4-6

6.2 CFWD Busy Number 4-6

6.3 CFWD No Ans Number 4-7

6.4 CFWD No Ans Delay 4-7

7 Time/Date 4-7

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8 Voice Mail 4-7 9 Network 4-8

9.1 DCHP 4-8

9.2 Current IP Address 4-8

9.3 Host Name 4-9

9.4 Domain 4-9

9.5 Current NetMask 4-9

9.6 Current Gateway 4-9

9.7 Enable Web Server 4-9

9.8 Non DHCP IP Address 4-10

9.9 Non DHCP Subnet Mask 4-10

9.10 Non DHCP Default Route 4-10

9.11 Non DHCP DNS 1 4-10

9.12 Non DHCP DNS 2 4-10

9.13 Non DHCP NTP Server 1 4-10

9.14 Non DHCP NTP Server 2 4-11

Contents

10 Product Info 4-11

10.1 Product Name 4-11

10.2 Serial Number 4-11

10.3 Software Version 4-11

10.4 Hardware Version 4-12

10.5 MAC Address 4-12

10.6 Client Cert 4-12

11 Status 4-12

Phone 4-12 Ext 1/2/3/4 4-13

Line 1, 2,3,4 4-13 12 Reboot 4-13 13 Restart 4-13 14 Factory Reset 4-13 15 Set Password 4-14 16 Set LCD Contrast 4-14 17 CallPark Status 4-14 18 Language (SPA922, 942, and 962) 4-14

CHAPTER

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5 SPA900 Series Phone Field Reference 4-1

Info Tab 4-2

System Information 4-2

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Contents

Product Information 4-2 Phone Status 4-3 Ext 1/2/3/4/5/6 Status 4-4 Line 1/2/3/4/5/6 Status 4-4 Downloaded Ring Tone 4-5

System Tab 4-6

System Configuration 4-6 Internet Connection Type 4-6 Static IP Settings 4-7 PPPoE Settings 4-7 Optional Network Configuration 4-7 VLAN Settings 4-8

SIP Tab 4-9

SIP Parameters 4-9 SIP Timer Values (sec) 4-11 Response Status Code Handling 4-12 RTP Parameters 4-13 SDP Payload Types 4-14

4-17

NAT Support Parameters 4-17 Linksys Key System Parameters 4-18

Regional Tab 4-19

Call Progress Tones 4-19 Distinctive Ring Patterns 4-21 Control Timer Values (sec) 4-22 Vertical Service Activation Codes 4-22 Outbound Call Codec Selection Codes 4-27 Miscellaneous 4-29

Phone Tab 4-33

General 4-33 Line Key 1/2/3/4/5/6 4-33 Miscellaneous Line Key Settings 4-34 Line Key LED Pattern 4-34 Supplementary Services 4-35 Ring Tone 4-36 Auto Input Gain (dB) 4-37 Background Picture (SPA 962) 4-37

Ext 1/2/3/4/5/6 Tab 4-38

General 4-38

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Share Line Appearance 4-38

NAT Settings 4-39

Network Settings 4-39

SIP Settings 4-40

Call Feature Settings 4-42

Proxy and Registration 4-43

Subscriber Information 4-44

Audio Configuration 4-46

Dial Plan 4-47 User 4-49

Call Forward 4-49

Speed Dial 4-50

Supplementary Services 4-50

Audio Volume 4-51

Phone GUI Menu Color Settings (SPA962) 4-51

Contents

APPENDIX

NDEX

A Acronyms

B Glossary

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Preface

This guide describes administration and use of the Linksys SPA900 Series IP phones. It contains the following sections:

• Document Audience, page xi

• Linksys 900 Series IP Telephones, page xi

• How This Document is Organized, pag e xii

• Document Conventions, page xii

• Related Documentation, page xiii

• Technical Support, page xiii

Document Audience

This document is written for the following audience:

• Service providers offering services using LVS products

• VARs and resellers who need LVS configuration references

• System administrators or anyone who performs LVS installation and administration

Note This guide does not provide the configuration information required by specific service

providers. Please consult with the service provider for specific service parameters.

Linksys 900 Series IP Telephones

The following summarizes the ports and features provided by the Linksys 900 Series IP phones described in this document.

• SPA901—One line, small, affordable, no display.

• SPA921—One-line business phone.

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• SPA922—One-line business phone with Power over Ethernet (PoE) support and an extra Ethernet

port for connecting another device to the LAN.

Linksys 900 Series IP Phone Administrator Guide

How This Document is Organized

• SPA941—Default two lines, upgradeable to four lines.

• SPA942—Default is two lines, upgradeable to four lines. Power over Ethernet (PoE) support and an

extra Ethernet port for connecting another device to the LAN.

• SPA962—Six lines, hi-res color display. Power over Ethernet (PoE) support and an extra Ethernet

port for connecting another device to the LAN.

Note PoE units (SPA922, SPA942, and SPA962) do not come with an external power adapter. The

PA100 power supply must be ordered separately if you are not using a PoE switch.

How This Document is Organized

This document is divided into the following chapters and appendices.

Chapter Contents

Chapter 1, “Introducing Linksys 900 Series IP Phones”

Chapter 2, “Getting Started” This chapter describes how to use the different administration and

Chapter 3, “Managing Linksys 900 Series IP Phones”

Chapter 5, “SPA900 Series Phone Field Reference”

Appendix A, “Acronyms ” This appendix provides the expansion of acronyms used in this

Appendix B, “Glossary” This appendix defines the terms used in this document.

This chapter introduces the Linksys 900 Series IP phones.

configuration tools provided for managing a Linksys 9 00 Series IP phone.

This chapter describes how to configure and monitor a Linksys 9 00 Series IP phone.

This chapter lists the function and usage for each field or parameter on the Linksys 900 Series IP phone administration web server pages.

document.

Preface

Document Conventions

The following are the typographic conventions used in this document.

Typographic Element Meaning

Boldface Indicates an option on a menu or a literal value to be entered in a field.

<parameter> Angle brackets (<>) are used to identify parameters that appear on the

Italic Indicates a variable that should be replaced with a literal value.

Monospaced Font Indicates code samples or system output.

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configuration pages of the 900 Series phone administration web server. The index at the end of this docume nt contains an alphabe tical listing of each parameter, hyperlinked to the appropriate table in

Series Phone Field Reference”

Chapter 5, “SPA900

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Technical Support

If you are an end user of LVS products and need technical support, contact the reseller or Internet telephony service provider (ITSP) that supplied the equipment.

Technical support contact information for authorized Linksys Voice System partners is as follows:

• LVS Phone Support (requires an authorized partner PIN)

888 333-0244 Hours: 4am-6pm PST, 7 days a week

• E-mail support

voipsupport@linksys.com

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Technical Support

Preface

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Overview

CHAPTER

Introducing Linksys 900 Series IP Phones

This guide describes the administration and use of Linksys analog telephone adapters (ATAs). This chapter introduces the functionality of the Linksys 900 Series IP phones and includes the following sections:

• Overview, page 1-2

• Feature Descriptions, page 1-8

• Technology Background, page 1-12

• Where to Go From Here, page 1-16

Table 1-1 summarizes the ports and features provided by the Linksys 900 Series IP pho nes

described in this document.

Table 1-1 Linksys SPA900 Series IP Phones

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Ethernet (LAN)

Product Name RJ-45

SPA901 One (1) One (1) Small, affordable, no display SPA921 One (1) One (1) One-line business phone SPA922 Two (2) One (1) Power over Ethernet (PoE) support SPA941 One (1) Four (4) Default is 2-lines active, upgradeable SPA942 Two (2) Four (4) Default is 2-lines active, upgradeable. Power

SPA962 Two (2) Six (6) Six lines, hi-res color display

Note PoE units (SPA922, SPA942, and SPA962) do not come with an external power adapter. The

PA100 power supply must be ordered separately if you are not using a PoE switch.

Voice Lines Additional Features/Notes

over Ethernet (PoE) support

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SPA900 Series Features

Chapter 1 Introducing Linksys 900 Series IP Phones

Figure 1-1 illustrates how the IP phones are connected in a VoIP network, including the

SPA3102, which acts as a SIP-PSTN gateway. As shown, the RTP400 and WRTP54G provide QoS-enabled IP routers in addition to two ports for connecting analog telephone devices.

Figure 1-1 Linksys SPA900 Series IP Phones in a VoIP Network

PSTN

Up to 4 FXO lines Local voicemail

SPA400 SIP-PSTN gateway

Switch

ISP

Internet

SPA901, 921, 922, 941, 942, 962

SPA900 Series Features

The following telephony features are provided by the different models of the SP A900 Series IP phones:

• Shared Line Appearance **

–

SPA901: Two Call Appearances Accessed Via Flash Key or Hook-Flash

–

SPA921 and SPA922: Two call appearances

–

SPA941 and SPA942: Four call appearances

–

SPA962: Six call appearances

• Line Status Indicators

FXS1

Fax/Analog Phones

SPA9000

ITSP

IP PBX

FXS2

1-2

• Call Hold

• Music on Hold **

• Call Waiting

• Outbound Caller ID Blocking

• Call Transfer - Attended and Blind

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• Call Conferencing

• Call Pick Up - Selective and Group **

• Call Park and UnPark **

• Call Swap

• Call Back on Busy

• Call Blocking - Anonymous and Selective

• Call Forwarding - Unconditional, No Answer, On Busy

• Hot Line and Warm Line Automatic Calling

• Call Logs (60 entries each): Made, Answered, and Missed Calls

• Do Not Disturb (callers hear line busy tone)

• URI (IP) Dialing Support (Vanity Numbers)

• Date and Time with Intelligent Daylight Savings Support

• Call Duration and Start Time Stored in Call Logs

• Ten-User Downloadable Ring Tones - Ring Tone Generator Free from www.linksys.com

• Speed Dialing

• Automatic Redial

• Configurable Dial/Numbering Plan Support - per Line

SPA900 Series Features

• Intercom **

• Group Paging **

• DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy

• Syslog, Debug, Report Generation, and Event Logging

• Secure Call Encrypted Voice Communication Support

• Built-in Web Server for Administration and Configuration with Multiple Security Levels

• Automated Provisioning, Multiple Methods. Up to 256-Bit Encryption: (HTTP, HTTPS, TFTP)

• Optionally Require Admin Password to Reset Unit to Factory Defaults

• NAT Traversal

• Set Preferred CODEC, Per Call, All Calls

• Call Return - Redial Last Caller

• Configurable Dial/Numbering Plan Support

• Support Linksys Voice System Automatic Configuration

** Feature requires support by SIP server

SPA901 Features

The SPA901 provides the following features that are not needed with the SPA900 Series IP phones that provide an LCD display:

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• Built-in Interactive Voice Response (IVR) system to check status and change configuration

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SPA900 Series Features

• Ringer and Handset Volume Controls

• Handset Input Gain Adjustment

SPA92x, SPA94x, and SPA962 Features

The SPA921, SPA922, SPA941, SPA942, and SPA962 prov ide an LCD display and addition al features that are not provided with the SPA901, including the following:

• Line Status Indicators: Active Line, Name, and Number

• Menu-Driven User Interface

• Digits Dialed with Number Auto-Completion

• Caller ID Name and Number and Outbound Caller ID Blockin g

• On-Hook Dialing

• Redial from Call Logs

• Personal Directory with Auto-dial (100 entries)

• On Hook Default Audio Configuration (Speakerphone and Headset)

• Called Number with Directory Name Matching

Chapter 1 Introducing Linksys 900 Series IP Phones

• Call Number using Name - Directory Matching or via Caller ID

• Subsequent Incoming Calls with Calling Name and Number

• Name and Identity (Text) Displayed at Start Up

• Distinctive Ringing Based on Calling and Called Number

Ensuring Voice Quality

Voice quality perceived by the subscribers of the IP Telephony service should be indistinguishable from that of the PSTN. Voice quality can be measured with such methods as Perceptual Speech Quality Measurement (PSQM), with a scale of 1–5, in which lower is better; and Mean Opinion Score (MOS), with a scale of 1–5, in which higher is better.

Table 1-2 displays speech quality metrics associated with various audio compression

algorithms.

Table 1-2 Speech Quality Metrics

Algorithm Bandwidth Complexity MOS Score

G.711 64 kbps Very low 4.5 G.726 16, 24, 32, 40 kbps Low 4.1 (32 kbps) G.729a 8 kbps Low–medium 4

1-4

G.729 8 kbps Medium 4 G.723.1 6.3, 5.3 kbps High 3.8

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Note SPA900 Series IP phones support all the above voice coding algorithms.

The following factors contribute to voice quality:

• Audio compression algorithm—Speech signals are sampled, quantized, and compressed before they

are packetized and transmitted to the other end. For IP Telephony, speech signals are usually sampled at 8000 samples per second with 12–16 bits per sample. The compression algorithm play s a large role in determining the voice quality of the reconstructed speech signal at the other end. SP A90 0 Series IP phones support the most popular aud io compression algorithms for IP Telephony: G.711 a-law and µ-law, G.726, G.729a, and G.723.1.

The encoder and decoder pair in a compression algorithm is known as a codec. The compression ratio of a codec is expressed in terms of the bit rate of the compressed speech. The lower the bit rate, the smaller the bandwidth required to transmit the audio packets. Although voice quality is usually lower with a lower bit rate, it is usually higher as the complex ity of the codec gets higher at the same bit rate.

• Silence suppression—SP A900 Series IP phones ap ply silence suppression so that silence packets are

not sent to the other end to conserve more transmission bandwidth. Instead, a noise level measurement can be sent periodically during silenc e suppresse d inte rvals so that the other end can generate artificial comfort noise that mimics the noise at the other end (using a CNG or comfort noise generator).

• Packet loss—Audio packets are transported by UDP, which does not guarantee the delivery of the

packets. Packets may be lost or contain errors that can lead to audio sample drop-outs and distortions and lower the perceived voice qual ity. SPA900 Series IP phones apply an e rror concealment algorithm to alleviate the effect of packet loss.

SPA900 Series Features

• Network jitter—The IP network can induce varying delay of received packets. The RTP receiver in

SPA900 Series IP phones keeps a reserve of samples to absorb the network jitter, instead of pl aying out all the samples as soon as they arrive. This reserve is known as a jitter buffer. The bigger the jitter buffer, the more jitter it can absorb, but this also introduces bigger delay. Therefore, the jitter buffer size should be kept to a relatively small size whenever possible. If jitter buffer size is too small, many late packets may be considered as lost and thus lowers the voice quality . SPA900 Series IP phones dynamically adjust the size of the jitter buffer according to the network conditions that exist during a call.

• Echo—Impedance mismatch between the telephone and the IP Telephony gateway phone port can

lead to near-end echo. SPA900 Series IP phones have a near-end echo canceller with at least 8 ms tail length to compensate for impedance match. SPA900 Series IP phones implement an echo suppressor with comfort noise generator (CNG) so that any residual echo is not noticeable.

• Hardware noise—Certain levels of noise can be coupled into the conversational audio signals

because of the hardware design. The source can be ambient noise or 60

Hz noise from the power

adaptor. The SPA900 Series hardware design minimizes noise coupling.

• End-to-end delay—End-to-end delay does not affect voice quality directly but is an important factor

in determining whether subscribers can interact normally in a conversation taking place over an IP network. A reasonable delay figure should be about 50–100 ms. End-to-end delay larger than 300

ms is unacceptable to most callers. SPA900 Series IP phones support end-to-end delays well

within acceptable thresholds.

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Feature Descriptions

SPA900 Series IP phones are full featured, fully programmable IP phones that can be custom provisioned within a wide range of configuration parameters. This chapter contains a high-level overview of features to provide a basic understanding of the feature breadth and capabilities of SPA900 Series IP phones.

• SIP Proxy Redundancy, page 1-8

• Supported Codecs, page 1-8

• Other Features, page 1-9

SIP Proxy Redundancy

In typical commercial IP Telephony deployments, all calls are established through a SIP proxy server. An average SIP proxy server may handle tens of thousands of subscribers. It is important that a backup server be available so that an active server can be temporarily switched out for maintenance. SPA900 Series IP phones support the use of backup SIP proxy servers so that service disruption should be nearly eliminated.

Chapter 1 Introducing Linksys 900 Series IP Phones

A simple way to support proxy redundancy is to configure a static list of SIP proxy servers in the SP A900 Series IP phone configuration profile, where the list is arranged in order of priority . The SPA900 Series IP phone attempts to contact the highest priority proxy server wh enever possible.

The dynamic nature of SIP message routing makes the use of a static list of proxy servers inadequate in some scenarios. In deployments where user agents are served by different domains, for instance, it would not be feasible to configure one static list of proxy servers per covered domain into every SPA900 Series IP phone. One solution to th is situation is through the use of DNS SR V records. SPA900 Series IP phones can be instructed to contact a SIP proxy server in a domain named in SIP messages. The SPA900 Series IP phone consults the DNS server to get a list of hosts in the given domain that provides SIP services. If an entry exists, the DNS server returns an SRV record that contains a list of SIP proxy servers for the domain, with their host names, priority, listening ports, and so on. The SPA900 Series IP phone tries to contact the list of hosts in the order of their stated priority.

If the SPA900 Series IP phone is currently using a lower priority proxy server, it periodically probes the higher priority proxy to see whether it is back on line, and attempts to switch back to the higher priority proxy whenever possible.

Supported Codecs

Negotiation of the optimal voice codec sometimes depends on the ability of SPA900 Series IP phone to “match” a codec name with the far-end device/gateway codec name. SPA900 Series IP phones allow the network administrator to individually name the various codecs that are supported such that the correct codec successfully negotiates with the far-end equipment.

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Chapter 1 Introducing Linksys 900 Series IP Phones

The administrator can select the low-bit-rate codec used for each line. G.711a and G.711u are always enabled.

Table 1-3 describes the codecs supported by the Linksys SPA900 Series IP

phones.

Table 1-3 Codecs Supported by Linksys SPA900 Series IP Phones

Codec (Voice Compression Algorithm)

G.711 (A-law and mµ-law) This very low complexity codec supports uncompressed 64

G.726 This low complexity codec supports compressed 16, 24, 32,

G.729A The ITU G.729 voice coding algorithm is used to compress

Feature Descriptions

Description

kbps digitized voice transmission at one through ten 5 ms voice frames per packet. This codec provides the highest voice quality and uses the most bandwidth of any of the available codecs.

and 40 kbps digitized voice transmission at one through ten 10 ms voice frames per packet. This codec provides high voice quality.

digitized speech. Linksys supports G.729. G.729A is a reduced complexity version of G.729. It requires about half the processing power to code G.729. The G.729 and G.729A bit streams are compatible and interoperable, but not identical.

G.723.1 SPA900 Series IP phones support the use of ITU G.723.1

audio codec at 6.4 kbps. Up to two channels of G .723.1 can be used simultaneously. For example, Line 1 and Line 2 can be using G.723.1 simultaneously, or Line 1 or Line 2 can initiate a three-way conference with both call legs using G.723.1.

When no static payload value is assigned per RFC 1890, SPA900 Series IP phones can support dynamic payloads for G.726.

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Feature Descriptions

Other Features

Chapter 1 Introducing Linksys 900 Series IP Phones

Table 1-4 summarizes the features provided by SPA900 Series IP Phones.

Table 1-4 Linksys ATA Features

Feature Description

Music On Hold On a connected call, SPA900 Series IP phones may place the remote

party on call. If the remote party indicates that they can still receive audio while the call is holding, the MOH server sends streaming audio.

Secure Calls A user (if enabled by service provider or administrator) has the option

to make an outbound call secure in the sense that the audio packets in both directions are encrypted.

Adjustable Audio Frames Per Packet

This feature allows the user to set the number of audio frames contained in one R TP packet. Packets can be adjusted to contain from 1–10 audio frames. Increasing the number of packets decreases the bandwidth utilized, but it also increases delay and may affect voice quality.

DTMF In-Band and Out-of-Band (RFC 2833) (SIP INFO *) SPA900 Series

IP phones may relay DTMF digits as out-of-band events to preserve the fidelity of the digits. This can enhance the reliability of DTMF transmission required by many IVR applications such as dial-up banking and airline information.

Call Progress Tone Generation

SPA900 Series IP phones have config urable call progress tones. Parameters for each type of tone may include number of frequency components, frequency and amplitude of each component, and cadence information.

Call Progress Tone Pass Through

Jitter Buffer—Dynamic (Adaptive)

This feature allows the user to hear the call progress tones (such as ringing) that are generated from the far-end network.

SPA900 Series IP phones can buffer incoming voice packets to minimize out-of-order packet arrival. This process is known as jitter buffering. The jitter buffer size proactively adjusts or adapts in size, depending on changing network conditions.

SPA900 Series IP phones have a Network Jitter Level control setting for each line of service. The jitter level decides how aggressively SPA900 Series IP phones try to shrink the jitter buffer over time to achieve a lower overall delay. If the jitter level is higher, it shrinks more gradually. If jitter level is lower, it shrinks more quickly.

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Table 1-4 Linksys ATA Features (continued)

Feature Description

Feature Descriptions

Voice Activity Detection with Silence Suppression and Comfort Noise Generation

Configurable Dial Plan with Interdigit Timers

Voice Activity Detection (VAD) with Silence Suppression is a means of increasing the number of calls supported by the network by reducing the required bidirectional bandwidth for a single call. VAD uses a very sophisticated algorithm to distinguish between speech and non-speech signals. Based on the current and past statistics, the VAD algorithm decides whether or not speech is present. If the VAD algorithm decides speech is not present, the silence suppression and comfort noise generation is activated. This is accomplished by removing and not transmitting the natural silence that occurs in normal two-way connection. The IP bandwidth is used only when someone is speaking. During the silent periods of a telephone call, additional bandwidth is available for other voice calls or data traffic because the silence packets are not being transmitted across the network. Comfort Noise Generation provides artificially-generated background white noise (sounds), designed to reassure callers that their calls are still connected during silent periods. If Comfort Noise Generation is not used, the caller may think the call has been disconnected because of the “dead silence” periods created by the VAD and Silence Suppression feature.

SPA900 Series IP phones have three configurable interdigit timers:

• Initial timeout (T)—Handset off hook; no digit pressed yet.

• Long timeout (L)—One or more digits pressed, more digits needed to

reach a valid number (as per the dial plan).

• Short timeout (S)—Current dialed number is valid, but more digits

would also lead to a valid number.

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Report Generation and Event Logging

SPA900 Series IP phones report a variety of status and error reports to assist service providers in diagnosing problems and evaluating the performance of their services. The information can be queried by an authorized agent, using HTTP with digested authentication, for instance. The information may be organized as an XML page or HTML page.

Syslog and Debug Server Records

SPA900 Series IP phones support detailed logging of all activities for further debugging. The debug information may be sent to a configured Syslog server. SPA900 Series IP phones provide configuration settings that determine the type of activity/events that should be logged, as for instance, a debug level setting.

Dynamic Payload When no static payload value is assigned per RFC 1890, SPA900

Series IP phones can support dynamic payloads for G.726.

Call Statistics and Reporting

The statistics collected by SPA900 Series IP phones during normal operation statistics are available in the Info tab. Line status is reported for each line (1 and 2). Each line maintains up to 2 calls: Call 1 and 2.

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Technology Background

This section provides background information about the technology and protocols used by the ATA. It includes the following topics:

• Session Initiation Protocol, page 1-12

• Using 900 Series Phones with a Firewall or Router, page 1-12

Session Initiation Protocol

Linksys 900 Series IP phones are implemented using open standards, such as Session Initiation Protocol (SIP), allowing interoperation with all ITSPs supporting SIP. SIP request for connection to another subscriber in the network. The requestor is called the user agent server (UAS), while the recipient is called the user agent client (UAC).

Figure 1-2 SIP Requests and Responses

Chapter 1 Introducing Linksys 900 Series IP Phones

Figure 1-2 illustrates a

SIP UA

SIP Proxy

RTP

SIP Proxy

SIP UA

In a SIP VoIP network, when the SIP proxy receives a request from a UAS for a connection and it does not know the location of the UAC, it forwards the message to another SIP proxy in the network. Once the UAC is located and the response is routed back to the UAS, a direct peer-to-peer session is established between the two UAs. The actual voice traffic is transmitted between UAs over dynamically assigned ports using the Real-time Protocol (RTP).

Using 900 Series Phones with a Firewall or Router

When using a 900 Series phone behind a firewall or router, make sure that the following ports are not blocked:

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• SIP ports—By default, UDP port 5060 and 5061

• RTP ports—16384 to 16482

If security is not a concern in your environment, you can consider disabling SPI, if this function exists on your firewall.

Network Address Translation

This section describes issues that arise when using the LVS system on a network behind a network address translation (NAT) device. It includes the following topics:

• NAT Overview, page 1-13

• NAT Types, page 1-14

• Simple Traversal of UDP Through NAT, page 1-14

• SIP-NAT Interoperation, page 1-15

NAT Overview

Technology Background

Network Address Translation (NAT) allows multiple devices to share the same public, routable, IP address for establishing connections over the Internet. NAT is typically performed by a router that forwards packets between the Internet and the internal, private network.

The association between a private address and port and a public address and port is called a NAT mapping. This mapping is maintained for a short period of time, th at varies from a few seconds to several minutes. The expiration time is extended whenever the mapping is used to send a packet from the source device.

The ITSP may support NAT mapping using a Session Border Controller (see Figure 1-3).

Figure 1-3 NAT Support with Session Border Controller Provided by ITSP

192.168.1.101

Private IP address

192.168.1.1

192.168.1.102

NAT Device

DHCP server

SPA9000

SIP Proxy

192.168.1.100

External IP address assigned by ISP

ISP

Internet

ITSP

Session Border

Controller

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Technology Background

NAT Types

Chapter 1 Introducing Linksys 900 Series IP Phones

This is the preferred option because it eliminates the need for managing NAT on the 900 Series phone. If this is not available, you need to discuss with the ITSP how to use the NAT Support Parameters provided by the 900 Series phone, such as <Outbound Proxy> and <STUN Server Enable>.

A typical application of a NAT is to allow all the devices in a subscriber home network to access the Internet through a router with a single public IP address assigned by an ISP. The IP header of the packets sent from the private network to the public network is substituted by NAT with the public IP address and a port assigned by the router. The receiver of the packets on the public network sees the packets as coming from the external address instead of the private address of the device.

The ways that NAT is implemented can be divided into the following categories:

• Full cone NAT—Also known as one-to-one NAT. All requests from the same int ernal IP address and

port are mapped to the same external IP address and port. An external host can send a packet to the internal host, by sending a packet to the mapped external address

• Restricted cone NAT—All requests from the same internal IP address and port are mapped to the

same external IP address and port. Unlike a full cone NAT, an external host can send a packet to the internal host only if the internal host had previously sent a packet to it.

• Port restricted cone NAT/symmetric NAT—Port restricted cone NAT or symmetric NAT is like a

restricted cone NAT, but the restriction includes port numbers. Specifically, an externa l host can send a packet to a particular port on the internal host only if the internal host had previously sent a packet from that port to the external host.

With symmetric NAT, all requests from the same internal IP address and port to a specific destination IP address and port are mapped to a unique external source IP address and port. If the same internal host sends a packet with the same source address and port to a different destination, a different mapping is used. Only an external host that receives a packet can send a UDP packet back to the internal host.

Simple Traversal of UDP Through NAT

Simple Traversal of UDP through NAT s (STUN) is a protocol defined by RFC 3489, that allows a client behind a NAT device to find out its public address, the type of NAT it is behind, and the port associated on the Internet connection with a particular local port. This information is used to set up UDP communication between two hosts that are both behind NAT routers. Open source STUN software can be obtained at the following website:

http://www.voip-info.org/wiki-Open+Source+VOIP+Software

STUN does not work with a symmetric NAT router. To determine the type of NAT your router uses, complete the following steps:

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Step 1 Enable debugging on the 900 Series phone:

1. Make sure you do no t have firewall runn ing on you r PC that could bl ock the syslog por t (by default

this is 514).

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2. On the administration web server , System tab, set <Debug Server> to the IP address and port number

of your syslog server. Note that this address and port number has to be reachable from the SPA900 Series IP phone.

3. Set <Debug level> to 3, but do not change the value of the <syslog server> parameter.

4. To capture SIP signaling messages, under the Line tab, set <SIP Debug Option> to Full. The output

is named syslog.514.log.

Step 2 To determine the type of NAT your router is using set <STUN Test Enable> to yes. Step 3 View the syslog messages to determine if your network uses symmetric NAT.

SIP-NAT Interoperation

In the case of SIP, the addresses where messages/data should be sent to a 900 Series phone system are embedded in the SIP messages sent by the device. If the 900 Series phone system is sitting behind a NAT device, the private IP address assigned to it is not usable for communications with the SIP entities outside the private network.

Technology Background

Note If the ITSP offers an outbound NAT-Aware proxy, this discovers the public IP address from the remote

endpoint and eliminates the need to modify the SIP message from the UAC.

The 900 Series phone system must substitute the private IP address information with the proper external IP address/port in the mapping chosen by the underlying NAT to communicate with a particular public peer address/port. For this, the 900 Series phone system must perform the following tasks:

• Discover the NAT mappings used to communicate with the peer.

This can be done with the help of an external device, such as a STUN server. A STUN server responds to a special NAT-Mapping-Discovery request by sending back a message to the source IP address/port of the request, where the message contains the source IP address/port of the original request. The 900 Series phone system can send this request when it first attempts to communicate with a SIP entity over the Internet. It then stores the mapping discovery results returned by the server.

• Communicate the NAT mapping information to the external SIP entities.

If the entity is a SIP Registrar, the information should be carried in the Contact header that overwrites the private address/port information. If the entity is another SIP UA when establishing a call, the information should be carried in the Contact header as well as in the SDP embedded in SIP message bodies. The VIA header in outbound SIP requests might also need to be substituted with the public address if the UAS relies on it to route back responses.

• Extend the discovered NAT mappings by sending keep-alive packets.

Because the mapping is alive only for a short perio d, the 900 Seri es phone system continu es to send periodic keep-alive packets through the mapping to extend its validity as necessary.

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Where to Go From Here

To do this ... Refer to

Use the different administration and configuration tools provided for managing Linksys 900 Series IP phone.

Configure and monitor a Linksys 900 Series IP phone.

Refer to the function and usage for each field or parameter on the Linksys 900 Series IP phone administration web server pages.

Find the expansion of an acronym used in this document.

Find the definition of a term used in this document.

Chapter 1 Introducing Linksys 900 Series IP Phones

Chapter 2, “Getting Started”

Chapter 3, “Managing Linksys 900 Series IP Phones”

Chapter 5, “SPA90 0 Series Phone Field Reference”

Appendix A, “Acronyms”

Appendix B, “Glossary”

The following documentation provides additional documentation for Linksys SPA900 Series IP phones:

• IVR Quick Guide

• SPA Provisioning Guide

The following documentation describes how to use other Linksys Voice System products:

• SPA9000 Administrator Guide

• LVS CTI Integration Guide

• LVS Integration with ITSP Hosted Voicemail Guide

• Linksys Voice over IP Product Guide: SIP CPE for Massive Scale Deployment

• SPA 2.0 Analog Telephone Adapter Administrator Guide

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CHAPTER

Getting Started

This chapter describes the tools and utilities available for administering Linksys SPA900 Series phones. It includes the following sections:

• Linksys 900 Series IP Phones, page 2-1

• Establishing Connectivity, page 2-9

• Using the Administration Web Server, page 2-11

• Web Interface URLs, page 2-13

• Provisioning, page 2-14

• Using the Interactive Voice Response Interface, page 2-16

Note If the SPA900 Series IP phone is supplied or sponsored by an Internet telephone service provider (ITSP),

certain network and service settings may be preconfigured. Depending on the configuration policy, access by an end user to specific configuration settings may be restricted or blocked.

Linksys 900 Series IP Phones

The Linksys SPA900 Series provides fully -featured VoIP phones that integrate with the Linksys SPA9000 to provide connectivity to other local station s, and through an ITSP to IP phones over the Internet, In addition, the optional SPA400 integrates with the SPA9000 and provides connectivity between SPA900 IP phones and the PSTN. This section summarizes the ports and hardware features provided by each device. It includes the following topics:

• Caring for Your Hardware, page 2-2

• SPA901 , page 2-2

• SPA92x , SPA94x, and SPA962 Hardware Feature s, page 2-3

• SPA922 , page 2-5

• SPA941 , page 2-5

• SPA942 , page 2-7

• SPA962 , page 2-7

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Linksys 900 Series IP Phones

Caring for Your Hardware

The Linksys 900 Series IP phones are electronic devices that should not be exposed to excessive heat, sun, cold or water. To clean th e equipment, use a slightly moistened paper or cloth towel. Do not spray or pour cleaning solution directly onto the hardware unit.

SPA901

The SPA901 provides an entry-level IP phone th at can be wall mounted (see Figure 2-1). The following are the hardware features provided by the SPA901:

• Voice Mail Message Waiting Indicator Light

• Redial Button

• Dedicated Flash Button

• Volume Control Button Cycles Through Volume Levels. Controls Ringer and Handset Volume.

• Standard 12-Button Dialing Pad

• High Quality Handset and Cradle

• Ethernet LAN – 10BaseT RJ-45

• 5-volt DC Universal (100-240 Volt) Switching Power Adaptor

Chapter 2 Getting Started

Figure 2-1 SPA901

The following tables describe the status indicators and controls on the front of the device and the ports on the back panel of the device.

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Front Panel and Side of Phone

Feature Function

Message waiting light On: you have a voicemail message. STATUS Off: make a call. On: make a call; line is shared. Flashing:

FLASH Press to pickup a second incoming call, transfer a call, or

REDIAL Press the redial the last number called.

Back Panel

Port Function

Phone jack Connect to the handset. Ethernet port Connect to the SPA9000 through a local switch. Power Connect to the 5-volt power supply.

Linksys 900 Series IP Phones

shared line on hold. On red: line in use. Flashing red: local call on hold. On orange: phone not registered.

setup a three-way conference call.

SPA92x, SPA94x, and SPA962 Hardware Features

The SPA921, SPA922, SPA941, SPA942, and SPA962 provide LCD displays and other additional hardware features, including the following:

• Four Illuminated Call Appearance Line Buttons with Tricolor LED's

• LED Indicates Line State – Active, Idle, On-Hold, Unregistered

• Line LED Configurable to 13 Different States (On/Off, Color, Flash)

• Dedicated Illuminated Buttons for:

• Audio Mute On/Off

• Headset On/Off

• Speakerphone On/Off

• Four Soft Key Buttons

• Four Way Rocking Directional Knob for Menu Navigation

• Voice Mail Message Waiting Indicator Light

• Voice Mail Message Retr ieval Bu tton

• Dedicated Hold Button

• Settings Button for Access to Feature, Set-up, and Configuration Menus

• Volume Control Rocking Up/Down Knob Controls Handset, Headset, Speaker, Ringer

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• Standard 12-Button Dialing Pad

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SPA921

Chapter 2 Getting Started

• High Quality Handset and Cradle

• Built-In High Quality Microphone and Speaker

• Headset Jack – 2.5 millimeter

• LED Test Function

• Two Ethernet LAN ports (100BaseT RJ-45)

• 802.3af Compliant Power over Ethernet (PoE)

• Optional 5 volt DC Universal (100-240 Volt) Switching Power Adaptor - Power Supply is Ordered

Separately

The SPA921 has one RJ-11 phone port and one 100BaseT RJ-45 port for connecting the phone to the LAN (see

Figure 2-2 SPA921

Figure 2-2).

Front Panel

2-4

The following tables describe the status indicators and controls on the front of the device and the ports on the back panel of the device.

Feature Function

LCD display Lists device status and configuration options. Telephone keypad Enters numeric digits for initiating a call or for entering

configuration information.

Navigation button Scrolls between display and configuration options in the LCD

display.

Soft keys 1-4 Selects options on the LCD display

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Back Panel

SPA922

Linksys 900 Series IP Phones

Port Function

Phone jack Connects to the handset. Ethernet ports Connects to the SP A9000 through a local switch. The SPA922

has two ports. Use the other port to connect to a PC or other LAN device.

Power Connect to the 5-volt power supply.

The SPA922 is similar to the SPA921, but provides Power over Ethernet (PoE) and an extra Ethernet port for connecting another device to the LAN (see supply must be ordered separately if you are not using a PoE switch.

Figure 2-3 SPA922

Figure 2-3). The SPA100 power

SPA941

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The SPA941 provides two lines b y default and is upgradeable to four lines (see Figure 2-4).

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Linksys 900 Series IP Phones

Figure 2-4 SPA941

The following tables describe the status indicators and controls on the front of the device and the ports on the back panel of the device.

Chapter 2 Getting Started

Front Panel

Back Panel

Feature Function

LCD display Lists device status and configuration options. Telephone keypad Enters numeric digits for initiating a call or for entering

configuration information.

Navigation button Scrolls between display and configuration options in the LCD

display. Soft keys 1-4 Selects options on the LCD display. Line status indicators 1-4 Displays status of each extension.

Port Function

Phone jack Connects to the handset. Ethernet ports Connects to the SP A9000 through a local switch. The SPA942

has two ports. Use the other port to connect to a PC or other

LAN device. Power Connects to the 5-volt power supply.

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SPA942

Linksys 900 Series IP Phones

The SPA942 is similar to the SPA941, but provides two Ethernet ports for connecting to the LAN and supports Power over Ethernet (see ordered separately if you are not using a PoE switch. See the table for SPA941.

Figure 2-5 SPA942

Figure 2-5). The PA100 power supply must be

SPA962

The SPA962 has a 320 x 240 true color, four-inch, LCD, provides up to six telephone extensions, and supports PoE (see

Figure 2-6). The PA100 po wer supply mu st be ordered

separately if you are not using a PoE switch.

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Linksys 900 Series IP Phones

Figure 2-6 SPA962

The following tables describe the status indicators and controls on the front of the device and the ports on the back panel of the device.

Chapter 2 Getting Started

Front Panel

Back Panel

Feature Function

Full-color LCD display Lists device status and configuration options. Telephone keypad Enters numeric digits for initiating a call or for entering

configuration information. Navigation button Scrolls between display and configuration options in the LCD

display. Soft keys 1-4 Selects options on the LCD display. Line status indicators 1-6 Displays status of each extension.

Port Function

Phone jack Connects to the handset. Ethernet ports (2) Connects to the SPA9000 through a local switch. Use the

second port to connect to a PC or other LAN device. Power Connects to the 5-volt power supply. AUX RJ-11 port Reserved for future functionality and not for use as a

telephone line port.

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Establishing Connectivity

This section describes how to connect the SPA900 Series IP phone hardware. It includes the following topics:

• Bandwidth Requirements, page 2-9

• Installing the SPA900 Series IP Phone, page 2-10

Bandwidth Requirements

Depending on how you have your IP phones configured, each call requires 55 to 110 kbps in each direction. Therefore, using G.729 as the voice codec setting, and with an average business-grade broadband Internet connection supporting 1.5 Mbps downstream and 384 kbps upstream, a total of seven (7) simultaneous conversations can be reliably supported with adequate bandwidth available for file downloads.

Linksys recommends using the 900 Series phone with QoS-capable networking equipment that can prioritize the VoIP application traffic. QoS features are available on many Linksys data networking switches and rout ers. A QoS-enabled router prioritizes the packets going upstream to the ISP.

Table 2-1 illustrates the bandwidth budget using different codecs.

Establishing Connectivity

Table 2-1 Ethernet Bandwidth Budget for Off-Net VoIP Calling

Approximate bandwidth budget for

Codec

each side of conversation

2 calls 4 calls 6 calls 8 calls

G.711 110 kbps 220 kbps 440 kbps 660 kbps 880 kbps G.726-40 87 kbps 174 kbps 348 kbps 522 kbps 696 kbps G.726-32 79 kbps 158 kbps 316 kbps 474 kbps 632 kbps G.726-24 71 kbps 142 kbps 284 kbps 426 kbps 568 kbps G.726-16 63 kbps 126 kbps 252 kbps 378 kbps 504 kbps G.729 55 kbps 110 kbps 220 kbps 330 kbps 440 kbps

This table is based on the following assumptions:

• Bandwidth Calculated with No Silence Suppression

• 20 Millisecond of payload per RTP packet

Note The use of silence suppression can reduce the average bandwidth budget by 30% or more.

For more information about bandwidth calculation, refer to the following webs ites:

http://www.erlang.com/calculator/lipb/

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http://www.packetizer.com/voip/diagnostics/bandcalc.html

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Establishing Connectivity

Installing the SPA900 Series IP Phone

This section describes how to install and verify operation of SPA900 Series IP phones. Check to make sure that you have the following package contents:

• Linksys 900 Series phone

• Ethernet cable

• Power adapter (must be ordered separately for PoE units: SPA922, SPA942, and SPA962)

Assembling the Phone and Connecting to the Network

Perform the following steps to connect 900 Series phone.

Step 1 Find the end of the coiled phone cord that has the longer straightened end. Plug that end into the back

of the phone base (phone symbol).

Step 2 Plug in the short straightened end of the handset cord into the jack at the bottom of the handset. Step 3 If you are using an external power source, push the power cord into the power supply, and plug the power

supply into the phone base unit. Use only the power supply that came with the phone. SPA922, SPA942 and SPA962 can be configured without external power, if connecting to an IEEE powered Ethernet switch. See Linksys.com for details.

Step 4 Plug the Ethernet cable into the back of the base station. Plug the other end of the Ethernet cable into

your already prepared network connection.

Chapter 2 Getting Started

If the phone is configured using an optional PoE adapter, see the PoE device documentation for setup instructions.

With the SPA922, SPA942, and SPA962, the second Ethernet port can be used to connect a PC. The AUX RJ-11 port is reserved for future functionality of the SPA962. It is not for use as a

telephone line port.

Note Do NOT plug a telephone line into the AUX port.

Attaching the Desk Stand

This and the following sections apply to a SPA900 Series IP phone with an LCD display. For information regarding the SPA901 IP phone, refer to the SPA901 QuickStart Guide.

Step 1 Line up the tabs on the desk stand with the slots on the back of the phone. Step 2 Slide the bottom tabs into the slots Step 3 Lightly press dow n on the top of the desk stan d. It should easily slide into the top slots. Do not force.

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Mounting the Phone to the Wall

Note When mounting the phone to a wall, do not attach the desk stand.

Step 1 Attach two appropriate screws to the wall. Leave 1/4” distance from the wall. This allows you to slide

the mounting brackets on to the screws.

Step 2 Push down slightly to lock the phone in place.

You can purchase an optional wall mount kit (MB100) from your retailer or Linksys.com .

Turning on the Phone

If you are using an external power source, instead of Power over Ethernet (PoE), plug the adapter to an electrical outlet. Use only the adapter supplied for the specific phone.

Using the Administration Web Server

• The phone executes a boot-up sequence when the power source is connected.

• During this sequence, the display first shows LINKSYS® or a customized message and then

“Initializing Network.”

• All of the lights on the phone flash.

• After the sequence, which should take about five se conds, the phone displ ays the standard menu for

the phone.

• The date, name, and number of the phone will display on the top line of the LCD screen.

At the bottom of the LCD screen, redial, dir, cfwd, and dnd are displayed over their associated soft buttons. A small arrow appears next to dnd indicating that more menus can be accessed by pushing the right side of the navigation button.

If the phone does not display the appropriate screen, verify the installation and connectivity. If these are correct, try unplugging the phone and plugging it back in again. If you still do not see the display, refer to the

“Troubleshooting FAQ” section on page 3-32.

Using the Administration Web Server

This section describes how to use the administration web server to configure the 900 Series phone. It includes the following topics:

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• Connecting to the Administration Web Server, page 2-12

• Administrator Account Privileges, page 2-12

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Using the Administration Web Server

Connecting to the Administration Web Server

To access the 900 Series phone administration web server, perform the following steps.

Step 1 Launch a web browser on a computer connected to the 900 Series phone . Step 2 Determine the address of the administration web server.

For a SPA900 Series phone with an LCD display, press Menu, 9, 2, or following these steps:

a. Press the Menu button. b. Scroll down to Network and press select. c. View the Current IP (Option 2).

For a SPA901 phone follo w these steps:

a. Press **** on the keypad to access the IVR menu. b. Press 110# to determine the Internet (WAN) IP address.

Step 3 Direct the browser to the IP address of the 900 Series phone.

Changing between the tabs on the Voice page does not discard the unsubmitted changes. You can wait until completing all your changes on the Voice pages before submitting th em.

Chapter 2 Getting Started

Note Changing between the Router and Voice pages discards any unsubmitted changes to either page.

Step 4 Click the Voice tab. Step 5 Click Admin and Advanced.

The Administrator account name is admin, and the User account name is user. These account names cannot be changed.

The system prompts for the Administrator account password if it has been set. If prompted, type the password provided by the ITSP and press Enter.

Step 6 To view the status information for the phone, click Status.

Enter the appropriate login information. Two views of the administration web server are available. Click basic to view basic settings, or click advanced to view advanced settings.

Step 7 If you make changes on a screen, click Submit All Changes to save the changes,

Click Undo All Changes button to undo your changes. When changes are saved, the 900 Series phone may reboot.

Administrator Account Privileges

The 900 Series phone support two levels of administration privileges: Administrator and User. Both privileges can be password protected.

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Note By default, there are no passwords assigned for either the Administrator account or the User account.

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Web Interface URLs

The Administrator account has the privilege to modify all the web profile parameters and can also modify the passwords of both Administrator and User account. The User account only has the privilege to access part of the web profile parameters. The parameters that the User account can access are specified using the Administrator account on the Provisioning page of the administration web server.

To directly access the Administrator account level privilege, use the following URL: http://ipaddress/admin/voice If the password has been set for the Administrator account, the browser prompts for

authentication. The User account name and the Administrator account name cannot be changed.

When browsing pages with the Administrator account privilege, you can switch to User account privilege by clicking the User Login link.

If the User account password is set, the browser prompts for authentication when you click the User Login link). From the User account, you can switch to the Administrator account by clicking the Admin Login link. Authentication is required if the Administrator account password has been set.

Note Switching between User and Administrator accounts or between basic and advanced views discards any

uncommitted changes that have already been made on the web pages.

Web Interface URLs

The 900 Series phone web interface supports several functions through sp ecial URLs:

• Upgrade

• Reboot

• Resync

Administrator account privilege is needed for these functions.

Upgrade URL

The Upgrade URL lets you upgrade the 900 Series phone to the firmware specified by the URL, which can identify either a TFTP or HTTP server.

Note If the value of the <Upgrade Enable> parameter in the Provisioning page is No, you cannot upg rade the

900 Series phone even if the web page indicates otherwise.

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The syntax of the Upgrade URL is as follows:

• http://spa-ip-addr/admin/upgrade?[protocol://][server-name[:port]][/firmware-pathname]

Both HTTP and TFTP are supported for the up grade operation.

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Provisioning

Resync URL

Note The SPA resyncs only when it is idle.

Chapter 2 Getting Started

If no protocol is specified, TFTP is assumed. If no server-name is specified, the host that requests the URL is used as server-name.

If no port specified, the default port of the protocol is used. (69 for TFTP or 80 for HTTP) The firmware-pathname is typically the file name of the binary located in a directory on the

TFTP or HTTP server. If no firmware-pathna me following example: http://192.168.2.217/admin/upgrade?tftp://192.168.2.251/spa.bin

The Resync URL lets you force the 900 Series phone to do a resync to a profile specified in the URL, which can identify either a TFTP, HTTP, or HTTPS server.

The syntax of the Resync URL is as follows: http://spa-ip-addr/admin/resync?[[protocol://][server-name[:port]]/profile-pathname]

is specified, /spa.bin is assumed, as in the

If no parameter follows /resync?, the Profile Rule setting from the Provisioning page is used. If no protocol is specified, TFTP is assumed. If no server-name is specified, the host that

requests the URL is used as server-name. If no port is specified, the default port is used (69 for TFTP, 80 for HTTP, and 443 for HTTPS). The profile-path is the path to the new profile with which to resync, for example: http://192.168.2.217admin/resync?tftp://192.168.2.251/spaconf.cfg

Reboot URL

The Reboot URL lets you reboot the 900 Series phone.

Note The 900 Series phone reboots only when it is idle.

The Reboot URL is http://spa-ip-addr/admin/reboot.

Provisioning

This section describes the provisioning functionality of the 900 Series phone. This section includes the following topics:

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• Provisioning Capabilities, page 2-15

• Configuration Profile, page 2-15

For detailed information about provisioning the 900 Series phone, refer to the Linksys SPA Provisioning Guide.

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Provisioning Capabilities

The 900 Series phone provides for secure provisioning and remote upgrade. Provisioning is achieved through configuration profiles transferred to the device via TFTP , HTTP, or HTTPS.

The 900 Series phone can be configured to automatically resync its internal configuration state to a remote profile periodically and on power up. The automatic resyncs are controlled by configuring the desired profile URL into the device.

The 900 Series phone accepts profiles in XML format, or alternatively in a proprietary binary format, which is generated by a profile compiler tool available from Linksys. The 900 Series phone supports up to 256-bit symmetric key encryption of profiles. For the initial transfer of the profile encryption key (initial provisioning stage), the 900 Series phone can receive a profile from an encrypted channel (HTTPS with client authentication), or it can resync to a binary profile generated by the Linksys-supplied profile compiler. In the latter case, the profile compiler can encrypt the profile specifically for the target 900 Series phone, without requiring an explicit key exchange.

Remote firmware upgrade is achieved via TFTP or HTTP (firmware upgrades using HTTPS are not supported). Remote upgrades are controlled by configuring the desired firmware image URL into the 900 Series phone via a remote profile resync.

Provisioning

For further information about remote provisio ning refer to th e LVS SPA Provisioning Guide.

Configuration Profile

The 900 Series phone configuration profile can be either an XML file or a binary file with a proprietary format.

The XML file consists of a series of elements (one per configuration parameter), encapsulated within the element tags <flat-profile> … </flat-profile>. The encapsulated elements specify values for individual parameters. The following is an example of a valid XML profile:

Binary format profiles contain 900 Series phone parameter values and user access permissions for the parameters. By convention, the profile uses the extension .cfg (for example, spa2000.cfg). The Linksys Profile Compiler (SPC) tool compiles a plain-text file containing parameter-value pairs into a properly formatted and encrypted .cfg file. The SPC tool is available from Linksys for the Win32 environment and Linux-i386-elf environment. Requests for SPC tools compiled on other platforms are evaluated on a case-by-case basis. Please contact your Linksys sales representative for further information about obtaining the SPC tool.

The syntax of the plain-text file accepted by the profile compiler is a series of parameter-value pairs, with the value in double quotes. Each parameter-value pair is followed by a semicolon. The following is an example of a valid text source profile for input to the SPC too l:

<flat-profile> <Admin_Passwd>some secret</Admin_Passwd> <Upgrade_Enable>Yes</Upgrade_Enable> </flat-profile>

Admin_Passwd “some secret”; Upgrade_Enable “Yes”;

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Using the Interactive Voice Response Interface

The names of parameters in XML profiles can generally be inferred from the 900 Series phone configuration web pages, by substituting underscores (_) for spaces and other control characters. Further, to distinguish between Lines 1, 2, 3, and 4, corresponding parameter names are augmented by the strings _1_, _2_, _3_, and _4_. For example, Line 1 Proxy is named Proxy_1_ in XML profiles.

Parameters in the case of source text files for the SPC tool are similarly named, except that to differentiate Line 1, 2, 3, and 4, the appended strings ([1], [2], [3], o r [4]) ar e used. For example, the Line 1 Proxy is named Proxy[1] in source text profiles for input to the SPC.

Using the Interactive Voice Response Interface

This section describes how to read or write basic network configuration settings using the keypad on the SPA901 IP Phone. To configure the other SPA phones, use the LCD display (see the

“Using the LCD Display” section on page 3-1) or the administration web server (see the

“Using the Administration Web Server” section on page 2-11.

This section includes the following topics:

Chapter 2 Getting Started

• Using the IVR Menu, page 2-16

• IVR Options, page 2-17

• Entering a Password through the IVR, page 2-19

By default, there is no password required for any of the IVR options. If the Administrator account password is set, password authentication is required for some options.

The interactive voice response (IVR) features that are available depend on your system configuration. A convenient quick-reference for the IVR is available at the following website:

http://www.linksys.com/

Using the IVR Menu

To use the IVR menu, complete the following steps.

Step 1 Connect an analog telephone to the Phone 1 or Phone 2 port of the 900 Series phone.

Note You can only access the IVR menu through an analog telephone, not an IP phone.

Step 2 Press **** (quickly press the star key four times).

Wait un til you hear “Lin ksys configuration menu.”

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Note You cannot access the IVR from a phone that is currently connected to a call.

Step 3 Refer to Table 2-2 to identify the option required. Step 4 Enter the required option followed by the # (pound) key.

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Step 5 To exit the menu, hang up the telephone.

Using the Interactive Voice Response Interface

To enter a period, use the star key (*). When entering a value, such as an IP address, to exit without entering any changes, press the

* (star) key twice within half a second. Otherwise, the * is treated as a decimal point. After entering a value, such as an IP address, press the # (pound) key to indicate you have

finished your selection. T o save the new setting, press 1. T o review the new setting, press 2. To re-enter the new setting, press 3. To cancel your entry and return to the main menu, press * (star).

For example, to enter the IP address 191.168.1.105 by keypad, press these keys: 191*168*1*105. Press the # (pound) key to indicate that you have finished entering the IP address. Then press 1 to save the IP address or press the * (star) key to cancel your entry and return to the main menu.

If the menu is inactive for more than one minute, the 900 Series phone times out. You need to re-enter the menu by pressing ****.

The settings you have saved take effect after you hang up the telephone. The 900 Series phone may reboot at this time.

IVR Options

Table 2-2 summarizes the options provided by the IVR.

Table 2-2 IVR Options

IVR Action IVR Menu Choice Parameter(s) Notes

Enter IVR Menu * * * * None Ignore SIT or other tones until you

hear, “Linksys configuration menu. Enter option followed by the pound

key or hang-up to exit.” Exit IVR Menu 3948 None Check DHCP 100 None IVR announces whether DHCP is

enabled or disabled. Enable/Disable DHCP 101 Enter 1 to enable

Enter 0 to disable

Check WAN IP

110 None IVR announces the current IP address

Address Set Static IP Address 111 Enter IP address using

numbers on the telephone key pad. Use the * (star) key when entering a decimal point.

Requires password

of the WAN port.

DHCP must be “Disabled,” otherwise

you hear, “Invalid Option,” if you try

to set this value.

Requires password

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Using the Interactive Voice Response Interface

Table 2-2 IVR Options (continued)

Check Network Mask 120 None IVR announces the current network

mask of SPA.

Set Network Mask 121 Enter value using numbers

on the telephone key pad. Use the * (star) key when entering a decimal point.

Check Static Gateway

130 None IVR announces the current gateway

IP Address Set Static Gateway IP

Address

131 Enter IP address using

numbers on the telephone key pad. Use the * (star) key when entering a decimal point.

DHCP must be “Disabled,” otherwise you hear, “Invalid Option,” if you try to set this value.

Requires password

IP address of SPA. DHCP must be “Disabled,” otherwise

you hear, “Invalid Option,” if you try to set this value.

Requires password

Check MAC Address 140 None IVR announces the MAC address o f

SPA in hex string format.

Check Firmware Version

Check Primary DNS Server Setting

Set Primary DNS Server

150 None IVR announces the v ersion of the

firmware running on the SPA.

160 None IVR announces the current setting in

the <Primary DNS> parameter.

161 Enter IP address using

Requires password numbers on the telephone key pad. Use the * (star) key when entering a decimal point.

Check administration web server port

170 None IVR announces the port that the web

server is listening on. (Default is 80)

Check LAN IP Address 210 None IVR announces the current IP address

of the LAN port.

Check PBX multicast

180 None IVR announces the current value.

address Set PBX multicast

address

181 Enter IP address and port.

Use * key for entering a

Enter a * between the IP address and

the Port fields. Requires password dot. For example,

224.168.168.169:8089 is 224*168*168*169*8089.

Enable/Disable administration web

7932 Enter 1 to enable

Enter 0 to disable

Requires password

server Manual Reboot of Unit 732668 None After you hear “Option Successful,”

hang up. Unit reboots automatically.

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Table 2-2 IVR Options (continued)

Using the Interactive Voice Response Interface

User Factory Reset of Unit

WARNING: ALL

“User-Changeable” NON-DEFAULT SETTINGS WILL BE LOST!

This might include network and service provider data.

Factory Reset of Unit WARNING: ALL NON-DEFAULT

SETTINGS WILL BE LOST!

This includes network and service provider data.

877778 Enter 1 to confirm

Enter *(star) to cancel operation

73738 Enter 1 to confirm

Enter * (star) to cancel operation

SPA prompts for confirmation. After confirming, you hear “Option Successful.” Hang up. Unit reboots and all “User Changeable” configuration parameters are reset to factory default values.

SPA prompts for confirmation. After confirming, you hear “Option Successful.” Hang up. Unit reboots and all configuration parameters are reset to factory default values.

Note The items marked with “Requires Password” require a password only if the Administrator password is

set.

Entering a Password through the IVR

To input the password using the phone keypad, the following translation convention applies:

–

To input: A, B, C, a, b, c—press “2’

–

To input: D, E, F, d, e, f—press “3’

–

To input: G, H, I, g, h, i—press “4’

–

To input: J, K, L, j, k, l— press “5’

–

To input: M, N, O, m, n, o—press “6’

–

To input: P, Q, R, S, p, q, r, s—press “7’

–

To input: T, U, V, t, u, v—press “8’

–

To input: W, X, Y, Z, w, x, y, z—press “9’

–

To input all other characters in the Administrator account password, press “0’

Note This translation convention applies only to the password input.

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Using the Interactive Voice Response Interface

For example, to input password test#@1234 by phone keypad, you need to press the following sequence of digits: 8378001234.

1. After entering a value, press the # (pound) key to indicate end of input.

–

To save value, press 1.

–

To review the value, press 2.

–

To re-enter the value, press 3.

–

To cancel the value entry and return to the main configurat ion menu , press *’ (star).

Notes:

–

The final # key is not included in the password value.

–

Saved settings take effect when the telephone is hung up, and if necessar y, the 900 Series phone automatically reboots.

2. After one minute of inactivity, the unit times out. The u ser needs to re-enter the configur ation menu

from the beginning by pressing * * * *.

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Managing Linksys 900 Series IP Phones

This chapter describes how to configure Linksys SPA900 Series IP phones and includes the following sections:

• Using the LCD Display, page 3-1

• Localization, page 3-6

• Changing the Display Background (SPA962), page 3-8

• Call Appearances and Extensions, page 3-9

• Line Key LEDs, page 3-10

• Using Call Features, page 3-11

• Configuring a Dial Plan, page 3-22

• System Administration, page 3-27

• Troubleshooting FAQ, page 3-32

Using the LCD Display

All SPA900 Series models provide an LCD display except for the SPA901, which you configure using the Interactive Voice Response (IVR) interface, described in the

Administration Web Server” section on page 2-11. This section describes the basic functions

of the LCD display and includes the following topics:

• LCD Display Controls, page 3-2

• Entering and Saving Settings, page 3-5

• Call Appearances and Extensions, page 3-9

• Using Soft Keys, page 3-3

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Using the LCD Display

LCD Display Controls

The LCD display on the SPA900 Series IP phones provides 18 menu options for operating, configuring and monitoring the IP phone. For information about each of these options, refer to

Chapter 4, “LCD Command Reference Guide.” Figure 3-1 illustrates the main controls on the

LCD display.

Figure 3-1 SPA900 Series LCD Controls

Chapter 3 Managing Linksys 900 Series IP Phones

Line status keys

Menu continuation indicator

Soft keys

Navigation rocker button

The function of the four soft keys depends on the option appearing on the LCD display. When more than four functions are available, a small left or right-facing triangle (menu continuation indicator) appears at the bottom of the LCD display . This indicates that more soft key functions can be accessed by pressing the Navigation button left or right.

To access an option, press the Menu button and press the Navigation button to select the option. Then push the left soft key beneath the Select option to select the option.

Figure 3-2

summarizes the function of the other buttons on the telephone keypad.

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Figure 3-2 Telephone Keypad

To use the shortcut to an option, press the Menu (Setup) button and type the number assigned to the option. The option does not have to appear on the LCD to use the shortcut. To select an option with a two-digit shortcut, press the second number within two seconds of the first number.

Using the LCD Display

Voicemail

Call hold

Menu (Setup) Microphone mute

Headset

Volume control

Speaker

Numeric keypad

Using Soft Keys

The LCD can display up to four options at one time. You can scroll through a longer list of options by pressing the Navigation button up or down. Pressing down or up when at the bottom or top of a list displays the first or last item in the list.

T ypically , when you select monitoring options, only the Cancel soft key is available. Table 3-1 summarizes the general soft key labels that appear on the LCD display.

Table 3-1 General LCD Display Soft Key Labels

Label Description

cfwd Forwards calls, with one of four options: All, When the line

dir Accesses the phone directory. dnd (Do not disturb) Sends incoming calls to voicemail, or gives

grPick Picks up a call from an extension outside the group. lcr (Last call return) Dials the last call received. park Parks a call at a designated call park number . The call is active

is busy , When there is no answer, or When there is no answer after a time delay.

a busy signal if voicemail is not available.

until it is unparked, or the caller disconnects (must be enabled before use).

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Table 3-1 General LCD Display Soft Key Labels

Label Description

pickup Picks up a call from another phone in the group. redial Displays a list of recently called numbers. unpark Picks up a call from where it was parked.

When selecting an option that allows modifying values, the LCD displays soft key labels indicating the functions available, such as add, paste, edit, and delete. To save changes, press the save soft key; to discard changes, press the cancel soft key.

Table 3-2 summarizes the function of each of the soft key labels that appear after pressing the

Menu (Setup) key.

Table 3-2 Menu (Setup) Soft Keys

Label Description

<< for more options to the left or

Moves left or right through entry without deleting characters.

>> for more options to the right

add Adds an entry. alpha, IP, num Used with directory and speed dial to enter a URL, IP address

or number. Press the soft key repeatedly to change selection.

cancel Cancels change before pressing save or ok. change Changes the ring tone feature. clear Clears the entire entry. copy In a directory, use the Navigation button to highlight an entry,

then press to copy an entry. delChr Deletes the last character entered. delete Deletes an entry. dial Places a call to the number highlighted on the LCD screen. edit Editspart of an entry. ok Confirms an operation. paste Use the Navigation button to highlight an entry in the

personal directory and press paste to place a copied entry

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play Play s ring tones. save Saves changes. select Chooses an item to change or view. y/n Toggles between yes and no settings.

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Entering and Saving Settings

The 900 Series phone GUI lets the user enter alphanumeric and numeric inputs. The following data types may be entered into the phone:

• Numeric string, such as a phone number when dialing

• Alphanumeric string, such a name when adding a directory entry

• IP address, such as a DNS server IP address

• Boolean (y/n), such as enabling or disabling the DHCP o pti on or sel ect ing a pr ef err ed au dio devi ce

or ring tone

When two or more numeric string, alphanumeric string, and IP address types can apply to an entry, the phone shows a soft key for switching input modes among the applicable types. The soft keys changes when pressed to allow the user to scroll through the available modes, such as num, alpha, or IP. The cursor also changes according to mode being used:

Table 3-3 Input Modes

Using the LCD Display

Table 3-3, to indicate the input

Input Mode Soft Key Cursor (appending) Cursor (inserting)

Numeric num Blinking underscore _ Blinking vertical line

Alphanumeric alpha Blinking full-height

block ¶

IP Address IP Blinking half-height

block .

Blinking open rectangle .

Blinking half rectangle .

When entering digits in numeric mode, enter the corresponding digits from the numeric key pad. When entering letters or symbols during alphanumeric mode, the phone shows a template of two or more choices as the user presses each digit key. The user scrolls through the choices by pressing the same key multiple times. To accept a symbol as input, the user does one of the following:

• Stop pressing the digit key for 1.5 seconds

• Pressing another digit key

• Navigate to another input field (if applicable) by pressing the UP/DOWN key, or

• Save the value by pressing the proper soft key, such save or ok

When entering an IP address, pressing the * digit inserts a dot (.), or a colon (:) if there are 3 dots entered already.

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When entering numeric, alphanumeric, or IP address values, the following additional soft keys are available for easier editing:

• <<—Move cursor to the left by one character

• >>—Move cursor to the right by one cha racter

• erase<—Erase the character to the left of the cursor

• clear—Clear the entire entry

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Localization

Chapter 3 Managing Linksys 900 Series IP Phones

Holding down the <<, >>, or erase<, soft key or any numeric keypad keys automatically repeats the entry. If the input value is wider than the LCD can display, the 900 Series phone automatically scrolls the input to the left so that the user can continue to enter input. On the other hand, the 900 Series phone stops accepting further input for a given field if the maximum allowed input length has been reached for that field. For Boolean input, the 900 Series phone displays (y/n); pressing the soft key toggles the current input between yes and no.

When the 900 Series phone displays an option, pressing the soft key either changes the input field to the next available choice in round-robin fashion, or displays a menu of choices. For most settings, the input value is saved once the user press save or ok before exiting the edit screen for that setting.

However, there are a few exceptions where the user must press save again when all the required changes have been made to the settings so that they are saved into the configuration at the same time. These exceptions are:

• Preferences

• Call Forward

• Network

• Speed Dialing

Localization

This section applies to the following SPA900 Series IP Phones:

• SPA922

• SPA942

• SPA962

This section describes how to localize the IP phone display language for these models, referred to in this section as SPA9x2 phones.

SPA9x2 phones support setting an alternative language on the user display. The administrator or service provider defines up to nine languages, in addition to English, to be made available and hosts the dictionaries for each of the languages on the HTTP or TFTP provisioning server . Language support follows Cisco dictionary principles.

The paragraphs describe parameters that can be defined in the SPA9x2 configuration file.

• Dictionary server script—This parameter is used to define the location of the dictionary server, the

languages available and the associated dictionary. The syntax is as follows:

<Dictionary_Server_Script ua="na"> </Dictionary_Server_Script>

The default is blank and the maximum number of characters is 512. The detailed format is as follows:

serv={server ip port and root path}; d0=<language0>;x0=<dictionary0 filename>; d1=<language1>;x1=<dictionary1 filename>; d2=<language2>;x2=<dictionary2 filename>; d3=<language3>;x3=<dictionary3 filename>; d4=<language4>;x4=<dictionary4 filename>;

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d5=<language5>;x5=<dictionary5 filename>; d6=<language6>;x6=<dictionary6 filename>; d7=<language3>;x7=<dictionary7 filename>; d8=<language8>;x8=<dictionary8 filename>; d9=<language5>;x9=<dictionary9 filename>;

The following is an example value:

<Dictionary_Server_Script ua="na"> serv=tftp://192.168.1.119/;d0=English;x0=enS_v101.xml;d1=Spanish;x1=esS_v101.xml </Dictionary_Server_Script>

• Language Selection—This parameter is used to select the default language. Its value needs to match

one of the languages supported by the dictionary server. The script (dx value) is as follows:

<Language_Selection ua="na"> </Language_Selection>

The default is blank and the maximum number of characters is 512. The following is an example:

<Language_Selection ua="na"> Spanish </Language_Selection>

These parameters can also be modified from the administration web server, on the Regional settings tab, as shown in Figure 3-3.

Localization

Figure 3-3 Dictionary Server Configuration Parameters

During startup, the phone checks the selected/preferred language and downloads the dictionary from the TFTP/HFTP provisioning server indicated in the device configuration. The dictionaries are available at the following website:

http://www.linksys-itsp.com The dictionaries are found in the FW and Tools section, LVS area. Currently dictionaries are

available for the following languages:

• English

• Spanish

• German,

• Dutch

• Italian

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• French

• Portuguese

• Danish

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Changing the Display Background (SPA962)

• Swedish

• Czech

• Slovak

The end user can change the language of the phone from the user menu, by following these steps:

1. Press the Menu button.

2. Select Language (option 18).

3. Select Edit to change lang uage.

4. Select Option to change the language.

5. Select OK and Save.

Changing the Display Background (SPA962)

This section applies only to the SPA962, which has a color LCD display with a customized background. You use any 320 x 240 pixel .BMP image for the background image that appears on the LCD display.

You can include the definition of th e required parameters in the configuration profile for the SPA962, or use the admin istration web server. The parameters are as follows:

• Picture download URL—The URL identifying the location of the .BMP file to be used as the

background image, including the TFTP server name (o r IP address), the directory, and the filename. For example:

tftp://myserver.mydomain.com/images/962back.bmp

The background image should be 320X240 pixels and mu st be in .BMP fo rmat.

• Current Background Picture—Select Default to use the default image; select Downloaded to use a

customized background image.

To use the administration web server to configure a customized image perform the following steps:

Step 1 Copy the 320X240 pixel .BMP image that you want to use to a TFTP server that is accessible from the

SPA962.

Step 2 Connect a browser to the administration web server for the SPA962. Step 3 Click the Phone tab and scroll down to the Background Picture section at the bottom of the page (see

Figure 3-4).

Figure 3-4 Background Picture Parameters

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Step 4 Select Downloaded from the Current Background Picture pull-down selection list.

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Step 5 Type the URL for the .BMP image, including the IP address or hostname of the TFTP server, the

directory, and filename in the Picture download URL field.

Step 6 Click Submit All Changes.

Call Appearances and Extensions

A client station is a SPA900 Series phone with one or more extension s and call appearances provisioned An extension in the SPA900 Series phone is a V oIP account in an ITSP network or an IP-PBX system. An extension can be identified with a unique user ID (typically, a phone number).

Up to four extensions can be configured on a SPA922 or 942 Series phone, while up to six extensions can be configured on a SPA962 phone. These are referred to as Ext n, or En, where n is the numeric identifier of the extension, such as Ext 1 or E1.

When the same extension is configured on more than one station, it is called a shared extension. Extension 1 is referred to as the primary extension. Some features can be activated only on the primary extension, such as Call Forwarding and Voice Mail Waiting Indicator (VMWI).

Call Appearances and Extensions

A call appearance physically corresponds to a line key on a station. There are four line keys on the SPA922 and 942 Series phones, and six line keys on the SPA962 phone. These are referred to as Line Key n, or Ln, where n is the numeric identifier of the extension, such as Line 1 or L1. Functionally, a call appearance is an instance of an extension. If an extension is assigned to line keys on multiple stations, it is called a shared line appearance.

One extension can be assigned to multiple line keys on a SP A900 Series phone. In fact, all four call appearances can be instances of the same extension. An extension is not a Shared Line Appearance unless it is also assigned to a line key on another station. Any of the four line keys can be disabled. Each call appearance supports one call at a time, either active or o n hold.

The VoIP Interface (VI) refers to an extension and its associated control parameters that are configured on a particular station. SPA900 Se ries phones include a rich set of configuration parameters to control the operation of the phone. Configuring an extension includes configuring the core account information and VI control parameters.

Althought the account information is usually the same for a shared extension on different stations, the rest of the VI parameters can differ. For example, the dial plan or the preferred codec to use when making a call on this extension could be different for two different stations sharing the extension.

A Call Appearance State refers to the state of a call appearance, which can be one of the following:

• Disabled: The line key is disabled

• Idle (Ready): The Call Appearance is ready for use

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• Dialing: Collecting digits from the user to be dialed out from this Line

• Calling: Waiting for the called party to respond

• Proceeding (a.k.a. Progressing): Called party station is ringing

• Ringing (a.k.a. Alerting): Incoming call, station is ringing

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Line Key LEDs

• Connected: Connected with remote party

• Held: Remote party is on hold

• Invalid: Remote party hangs up or error while attempting outbound call

• Busy: the line is being used by anot her station (shared l ine only )

A Call State is the Call Appearance State applied to a specific call instance. Call States are divided into two general categories:

• Active (Call) State: If the state is Dialing, Calling, Proceeding, Connected, or Invalid. When the

• Standby (Call) State: the state is Ringing or Holding. When the state of the Call is Standby, it is

Line Key LEDs

This section summarizes the behavior of the line key LEDS on SPA900 Series phones with multiple line key status indicators

Chapter 3 Managing Linksys 900 Series IP Phones

state of the Call is Active, it is referre d to as an Ac tive Call

referred to as a Standby Call

A line key corresponds to a call appearance. Call appearances for the same extension are numbered in ascending order of their line key position starting at 1. The line key LED color and blinking pattern are fully programmable.

The line key LED can be in one of four conditions:

• Red (r)

• Green (g)

• Amber (a)

• Off (o).

The 900 Series phone has four built- in blinking patterns defined:

• No Blink (nb): Steady on (or off)

• Slow Blink (sb): 500ms on, 500ms off

• Fast Blink (fb): 100ms on, 100ms off

• Double Blink (db): 100ms on, 100ms off, 100ms on, 700ms off

The administrator can also define arbitrary blinking pattern by modifying the syntax of an LedScript. The administrator can specify a different color and pattern for each of the following states of the call appearance.

• Idle: This call appearance is not in use. The user can use it to make outbound call on this station

• Local Seized: This call appearance has been seized by this station to prepare for an outbound call

• Local Progressing: This station is making an outbound call that is progressing

• Local Active: This station is engaged in a connected call on this call appearance

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• Local Ringing: This station is ringing for an incoming call on this call appearance

• Local Held: This station has placed this call appearance on hold

• Remote Seized: This call appearance has been seized by another station to prepare for an outbound

call

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• Remote Progressing: Another station is making a call on this call appearance and is progressing

• Remote Active: Another station is engaged in a connected call on this call appearance

• Remote Ringing: Another station is ringing for an incoming call to this call appearance

• Remote Held: Another station has placed this call appearance on hold

• Remote Undefined: The share call state is not known (this station is waiting for a notification from

the application server)

• Registration Failed: This station has failed to register with the proxy server for the corresponding

extension

• Registering: The station is attempting registration with the proxy server for the corresponding

extension.

• Disabled: This line key on this station is disabled

• Call Back: A call back (repeat dialing) operation is currently active on this call appearance

Using Call Features

This section describes how the main call features are used or configured. It includes the following topics:

• Selecting the Audio I/O Device and Line, page 3-12

• Making Calls, page 3-12

• Answering and Ending Calls, page 3-13

• Hold and Resume, page 3-13

• Call Waiting, page 3-13

• Speed Dialing, page 3-14

• Three-Way Conferencing, page 3-14

• Attended Call Transfer, page 3-15

• Blind Call Transfer, page 3-15

• Call Back, page 3-16

• Message Waiting Indication (MWI), page 3-16

• Accessing Voicemail, page 3-16

• Muting Calls, page 3-16

• Shared Call Appearances, page 3-17

• Personal Directory, page 3-17

• Caller and Called Name Matching , page 3-17

• Dialing Assistance, page 3-18

• Supplementary Services, page 3-18

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• Call Logs, page 3-18

• Audio Volume Adjustm ent, page 3-20

• Managing Ring Tones, page 3-20

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Selecting the Audio I/O Device and Line

The SPA900 Series IP phon es provide three audio I/O d evices:

• Handset

• Built-in microphone and spe aker

• External microphone and headse t

Only one audio device can be selected at one time. The speaker is selected by pressing the Speaker button, while the headset is selected by pressing the Headset button. When Speaker or Headset is selected, the corresponding LED is steady green.

The handset is selected whenever it goes from the on-hook to the off-hook position. Selecting or turning off an audio device is equivalent to an off-hook action, while turning on an audio device is equivalent to an on-hook

When there are no active calls, all audio devices are deselected. When any one of the calls becomes active, the handset is selected automatically as the audio device if it is off-hook, else the speaker or the headset is selected according to configuration: the user can configure whether speaker or the headset device has higher preference.]

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Making Calls

A line is selected by pressing the corresponding line key. To make or receive calls, the user must select a call appearance (line key) or an audio device. If the user selects a call appearance, then an audio device is selected automatically according to user preference setting. If the user selects an audio device, an idle call appearance is selected automatically in the order L1, L2, L3, and L4. The user may switch between audio devices while the call in any of the active states.

A new call appearance is selected automatically when the user presses the conf or xfer soft key . The 900

Series phone tries to pick an idle call appearance that is on the same extension as the last active call. If that call appearance is not available, the phone tries to pick a call appearance whose extension has the same proxy server as the last active call. Finally, if that fails, the first idle call appearance is selected, in numeric order (L1, L2, L3, L4).

There are two steps in making a call:

1. Selects an audio device

2. Dials the number

The 900 Series phone lets the user perform these two steps in either order. In other words, the user can select an audio device either before or after dialing. The 900 Series phone supports two types of dialing: explicit or implicit. Explicit dialing occurs when the user performs one of the following actions:

3-12

• Enters the target number digits one by one

• Enters the speed dial assignment of the target number

• Selects an entry from a directory, and presses the dial soft key

• Presses a dedicated key, such as the Voice Mail key

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Implicit dialing refers to dialing a highlighted entry from one of the device directories or logs. A number can be implicitly dialing by selecting an audio device while the number is highlighted on the display.

If an audio device is selected prior to dialing, only explicit dialing is supported. Both explicit and implicit dialing are allowed if the audio device is selected after dialing. Explicit dialing before the audio device is selected is called on-hook dialing.

When an audio device is selected before dialing, a call appearance (for making the call) is also indirectly selected. The 900 Series phone picks the first available call appearance, in descending order, starting with Line Key One. If no call appearances are available, no audio devices are selected. A call appearance can be explicitly selected by pressing the corresponding line key. An audio device is selected automatically based on the device programmed preferences.

Answering and Ending Calls

A line key LED rapidly blinks red when a call is incoming. The user can answer the call by:

• Selecting an audio device.

The 900 Series phone answers the ringing call. If there are simultaneous incoming calls, the call ringing on the lowest numbered line key is answered.

• Pressing the corresponding line key. The default audio device is automatically selected.

Using Call Features

When a call is in the active state (not on hold or parked), the call can be ended by turning off the currently selected audio device. A call in a standby state cannot be ended; it must be brought back to the active state first.

Hold and Resume

An active call can be placed on hold by explicitly pressing the Hold key, or by pressing another line key (to answer an incoming all, resume a held call, or start a new call). When a call is on hold, the corresponding line key LED slowly blinks red. T o resume a call that is on hold, press the corresponding line key. When the phone is hosting a three-way conference, two call appearances are in the connected state. Pressing the Hold key places both calls on hold. The calls can be individually brought back to the connected state by pressing the corresponding line keys.

Call Waiting

The Call Waitin g function is activated when a device has a call in the active state and another call is incoming. The phones in the SPA90 0 Series do not support multiple calls on the same line key . Incoming calls are assigned to an unused line key , causing the line key to quickly blink red. (Note that the Voice Mail Waiting Indicator also blinks red whenever there is an incoming call.) The phone does not ring. However, to alert the user, the call waiting tone is played into

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Speed Dialing

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the active audio device. The 900 Series phone allow multiple lines to simultaneously cause Call Waiting. All four call appearances can ring at the same time, after one of the extension keys is answered, the other three call appearances go into the Call Waiting state.

The 900 Series phone has eight programmable speed dial numbers. These use the administration web server User tab, or the LCD on phones that provide an LCD display.

To use the LCD to configure speed dialing, press the Menu button and then press the Speed Dial soft key. The speed dial numbers are assigned to the digits two through nine. If a slot is not configured, the LCD displays Not Assigned. Highlight the speed dial entry you want to add or modify, press edit to make the changes, and press ok to store the value.

A speed dial entry can be a phone number or URL. The user can also enter a name that matches one of the directory entries. As the value is entered, the 900 Series phone attempts to match the entry to the directory entries and brings up a list of potential matches on the screen. The user can scroll down to select one of the suggested entries. To dial a speed dial entry, dial the corresponding single digit like a 1-digit phone number. The user can enter # after the speed dial number but it is not necessary.

Three-Way Conferencing

There are two ways to initiate a three-way conference on the 900 Series phone. First, during an active call, if the phone has one or more idle call appearances, press the conf soft key. This places the active call on hold and selects the next available call appearance for dialing (the dial tone is heard in the active audio device). Dial the third party, the party to be added to the conference call. After the third party line rings or is answered, presses the conf soft key again to join the two calls into a conference.

Second, if the phone has a call on hold while another call is either ringing or is connected, the confLx soft key appears. If the phone has only one call on hold, pressing the confLx soft key joins the call on hold with the active call. If the phone has more than one call on hold, pressing the confLx soft key places the active call on hold and prompts the user to select the other line key to join with the conference call.

Once the conference starts, the 900 Series phone plays a special brief tone to all th ree parties to indicate that a conference call is in progress. The initiator of the conference call can terminate it at any time by hanging up. The initiator leave the conference without ending the call by pressing the join soft key; this allows the other two parties to continue the call. This is implemented as a call transfer, which may not work if the other two parties are hosted by different service providers.

If the extension is configured with an external Conference Bridge URL, the conference is not limited to three participants; it can include as many as the conference bridge supports. There are two restrictions that apply when using an external conference bridge:

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• The conference can only be initiated (by pressing the conf soft key) when the third party has

answered, instead of when the phone is either ringing or answered

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• Ringing call cannot be brought into a conference by pressing the confLx soft key; only calls that are

on hold can be brought into the conference with the confLx soft key.

Attended Call Transfer

An attended call transfer allows a user to optionally transfer a call to a third party, after having a discussion with that third party. There are two ways to perform an attended call transfer:

• During an active call, press the xfer soft key to place the current call on hold and to act ivate an idle

line key.

• Dial the transfer target number on the newly activated line. When the target is either ringing or

answered, press the xfer soft key again to complete the transfer.

While one or more call appearances is on hold, initiate a call on an idle line key. While the call is either ringing or is connected, press the xferLx line key . If only one other call is on hold, the active call is connected with the call on hold. Otherwise, the user is prompted to select to which line key to transfer the call.

At the completion of call transfer operation, the holding call peer is the transferee, and the connected or proceeding call peer is the transfer target. The case where a call transfer is completed when the transfer target is ringing is called a semi-attended call transfer. This is different from a blind transfer, which is described in the next subsection.

Using Call Features

Blind Call Transfer

A blind call transfer allows the user to transfer a call without speaking with the receiver of the transferred call. For the user, the call ends as soon as the transfer target phone number is dialed. The user phone sends the contact information for the transfer target to the phone of the other party on the call. The information is carried in a signaling message. When the message is sent, the user phone is dropped from the call and the other party phone automatically dials the transfer target.

To perform a blind call transfer, press the bxfer soft key and then dial the target number. The bxfer soft key automatically becomes active during a call, but it may not be visible on the display. If the bxfer key is not visible, press the left/right arrows on the Navigation button until it scrolls onto the display.

Some SPA phones do not have bxfer soft keys. To perform a blind transfer on these devices, during the call, press the xfer soft key, or the line key of an unused line. There must be a spare line available to perform the transfer. Once the new dial tone is heard, enter *98. The phone prompts the user to enter the phone number of the transfer target. Enter the number to complete the blind transfer. If the user phone has more than one call on-hold on, the last call placed on hold is the one transferred.

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Call Back

Chapter 3 Managing Linksys 900 Series IP Phones

This service mimics the call back service offered by the PSTN. The user can activate this service on a busy number such that he is called back as soon as the busy number becomes available. The 900 Series phone implements this feature by repeatedly dialing the busy number periodically until the called party rings or answers, or until the service order is canceled by the user or expires. For that reason, it is referred to as the repeat dialing service.

To activate this service, the user selects a line key and enters *66 (call back activation code). The 900 period and expiration time are configurable. To cancel the service, the user picks up any line and enters *86. When call back service is active, the corresponding Line key flashes green. The user can still use this line to make or answer calls. The call back service is temporarily paused until the call appearance is idle again.

When the call back number rings or answers, the call appearance rings like a normal incoming call (the ring to be used is the default ring for the corresponding extension for that Line key). If the call back party answers the call before the local user does, the 900 Series phone sends a holding tone to the call back party. If the local user picks up the call back Line key first, the user hears a ring back tone like a regular outbound call.

Series phone then uses this call appearance to call the last called number. The retry

Message Waiting Indication (MWI)

The 900 Series phone indicates message waiting in two ways:

• Turning on the red MWI light - LED (with an envelope icon right above LED) on the phone.

• Playing stutter tone instead of regular dial tone.

The MWI light only applies to new message indication received on Extension 1. It is not turned on or off due to new message indication received on the other extensions [2 to 4]. On the other hand, the 900 appearance according to the message waiting state of the corresponding Extension X.

Series phone can be configured to play a stutter tone on the individual call

Accessing Voicemail

The 900 Series phone has a voicemail retrieval key for quick access to voicemail. If a voicemail access number or URL is not specified, pressing the key brings up a screen where the user can enter the appropriate information. This information can also be entered from the web page or downloaded via provisioning. The 900 Series phone attempts to call the voicemail number or URL if there is an idle call appearance available.

Muting Calls

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The user can mute the audio input into the phone when an audio device is switched on, by pressing the MUTE key. To un-mute, press the MUTE key again. If no audio device is on, pressing the Mute key has no effect. When switching from the speaker phone to the handset, the phone is automatically un-muted.

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Shared Call Appearances

The 900 Series phone supports shared call appearances in association with a Broadsoft application server. An extension can be shared by two or more stations. All call appearances on a shared extension are shared. At any given time, each station sharing a call appearance can monitor the state of the call appearance. A station can select a share call appearance to make a call only if the call appearance is not being used by another station. All stations ring on an inbound call to the shared call appearance extension. Whoever picks up the call first takes the call. When a call is placed on hold by one station, it can be resumed from another station sharing the same call appearance.

Personal Directory

A directory entry consists of the following information:

• Name

• Phone number or (SIP) URL

• Ring Tone

Using Call Features

Up to 100 entries can be stored in the 900 Series phone. An entry can be added or edited from the phone GUI or on the web page or through remote provisioning. To do this from the web page or through remote provisioning, you specify the value as a DirEntry script. T o view or edit the personal directory via the phone web interface, use the link http://ip_address/pdir.htm The user can invoke the Directory menu on the phone in 2 ways:

• Press the Menu button and select Directory

• Press dir when not already in a Menu display

The user can add a new entry to the Directory by selecting the New Entry option (the first item) on the Directory menu and press add. To modify or delete an existing item, highlight the corresponding entry and press edit or del. When trying to delete an entry, the phone prompts the user for confirmation before proceeding.

The user can copy an existing directory entry to create a new entry. To do this, high light the entry you want to copy and press copy. Then highlig ht th e New Entry o ption and press paste. The user can edit the new entry and then press save to save the entry. The user can also save a call log entry into the directory . To do this, select a call log entry and press save. Then edit the entry if needed, and press save to save the new entry in the directory.

Caller and Called Name Matching

When making an outgoing call, the 900 Series phone tries to find the dialed number in the personal directory first, then from the missed call log, and finally from the answered call log. If a match is found and the name field is present in the matched entry, it is shown on the call screen as the called peer name. For incoming calls, 900 Series phone tries to find the caller phone number in the personal directory . If a matc h is found and the name field is present in the matched entry, it replaces the current caller ID name and is shown on the call screen as the

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Using Call Features

calling peer name. The same name also goes into the incoming call log. If a match is not found, or the name field is not present in the matched entry, the current caller ID name is used if it exists.

Dialing Assistance

Dialing assistance is an option the user can enable or disable under the Preferences menu. If the option is enabled, the phone shows up to ten potential matches from the Redial List (most recent first) and from the Personal Directory as the user dials the target number. This feature applies to both off-hook (with dial tone) and on-hook (without dial tone) dialing. When the suggestion list appears, the user can scroll down the list using the Up/Down key to select the desired target number from the list. When looking for matches from the Redial List and the Directory , the 900 Series phone skips any leading *codes, so that the match is purely based on the target number. If configured on the phone, the *code that is skipped may be any of the following:

• Call forward activation and deactivation codes

• Secure call activation and deactivation codes

Chapter 3 Managing Linksys 900 Series IP Phones

• Block CID activation and deactivation codes

• Block anonymous call activation and deactivation codes

• DND activation and deactivation codes

• Prefer codec and Force codec codes

• Referral services codes and Feature Dial Services codes

Supplementary Services

The 900 Series phone supports the following supplementary services:

• Block Caller ID —If enabled, the 900 Series phone attempts to hide the caller-ID on all outbound

calls by default. The user can also enable or disable this feature on a pe r cal l basis by pre dia ling a *code before making the call.

• Block Anonymous Call —Rejects all inbound calls with no caller-ID with a 406 response

• Do Not Disturb (DND) —Rejects all inbound calls as if the phone is busy. That is, the phone either

responds a 302 if call forward on busy if enabled, or 486 otherwise.

• Secure Call —If enabled, the 900 Series phone tries to negotiate on all outbound calls to use

encrypted media (SRTP) by default. The user can also enable or disable this feature on a per-call basis by pre dialing a *code before making the call. Note that to use Secure Call on an extension, <Mini Certificate> and <SRTP Private Key> must be configured for that extension. The enabling and disabling of these services applies to calls on all configured extensions on the 900

Series phone

Call Logs

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There are three call logs maintained for each VoIP interface:

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• Redial List—Each Redial List entry is added when dialing is completed, regardless if the number is

correct or not, or if the call is successful or not.

• Answered Calls—An Answered Call is logged when the incoming call is answered.

• Missed Calls—A Missed Call is logged for each incoming call that causes the phone to ring but not

answered.

The 900 Series phone keeps up to 60 entries per log in reverse chronolog ical order. The logs are saved in the phone non-volatile memory. All call logs can be viewed on the 900 Series phone web page using the following link:

http://ip_address./calllog.htm Each log is shown on the web page with the following syntax:

[ext-id]name, phone, time-stamp

Where:

• ext-id is the extension on which the call is made from or received at (E1, E2, E3, or E4). If [En] is

not present, it implies [E1].

• name is the call peer name. If thi s field is mi ssing, the peer nam e is not ava ilable.

• phone is the call peer phone number (user-id). If this field is missing, the peer phone number is not

available.

Using Call Features

• t is the time-stamp at which the call log is entered. It has the format mm/dd hh/mm<a|p> (a = am

and p = pm).

For example:

[E2]Joe Smith,14089991234,t=10/12 11:12a

Another example, when the name is not available:

14089991234,10/12 01:12p

Table 3-4 summarizes operations the user can perform with the call logs.

Table 3-4 Call Log Operations

Operation Steps

To view a call log

To delete a log

To edit a log

1. Press Menu.

2. Select Call History.

3. Select the log you want to view.

1. Highlight the entry.

2. Press del.

1. Highlight the entry.

2. Press edit.

3. Press save to save the changes, or cancel to abort the change.

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Table 3-4 Call Log Operations

Operation Steps

To call a log entry

To save a log entry into Personal Directory (if the directory is not already full)

Audio Volume Adjustment

There are four kinds of audio volume that can be adjusted:

• Ringer volume (on the speaker)

• Speaker

• Handset

• Headset

Chapter 3 Managing Linksys 900 Series IP Phones

1. Highlight the entry.

2. Press dial, or Off-Hook, or Turn on Speaker, or Turn on

Headset, or press a line key.

1. Highlight the entry.

2. Modify the entry as needed.

3. Press save to save the entry in the personal directory.

T o adjust the volume, press the VOL key to invoke the volume adjustment screen. Then use the UP/DOWN keys to increase or decrease the volume. The kind of volume to adjust is by context. When no audio device is selected at the moment, the Ringer volume is adjusted. Each UP/DOWN key press plays a short burst of the current ring tone. If one of the audio devices is selected, the volume adjustment applies to that device. The screen shows a pictorial representation of the volume level.

Each volume has 16 steps in nonlinear logarithmic scale. Each setting is a 16-bit multiplier in the range 41 to 32764 in Q3.12 format (or 0.01 to 7.999 or 40 to 18 dB). Each step is equal to 58/(16-1) or 3.87 dB.

Additional volume settings can be added and accessed via the phone setup screens. These volumes are:

• DTMF Echo Volume (use <DTMF Playback Level>)

• Call Progress Tone Volume (setting the ToneScript for each tone)

Managing Ring Tones

The 900 Series phone offers 10 ring tones. The characteristics of each ring tone is configurable using a Ring Tone script. In a Ring Tone script, the user can assign a name for the ring tone, and specify two components:

• Waveform—1, 2, 3, or 4

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• Cadence—1, 2, 3, or 8

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The user can assign a default ring tone for each extension. The user can assign a ring tone for each personal directory entry (which overrides the default ring tone). Ring tone can be selected from the user interface. In addition, the 900 Series phone allows two user-downloadable ring tones. Therefore a total of 12 ring tone choices are available as the default ring for an extension or assigned to individual callers in the personal directory.

User-downloaded ring tone are labeled User 1 and User 2 in the choices for <Default Ring>. On the phone ring tone menu, the User 1 and 2 choices are replaced by the corresponding name of the ring tone or Not Installed if the user ring tone slots have not been used.

The user ring tone (1 or 2) can be downloaded using the phone web interface with the link: http://phone_ip_addr/ringtone1[?<url>] Where <url> syntax is [tftp://][host[:port]]/<path>. Only TFTP is supported. If the host is not specified, the TFTP host is the web client. The

default port is 69. To remove the User 1 ring tone from the phone, set the <path> to delete, as follows:

http://phone_ip_addr/ringtone1?/delete The link is case sensitive.

Using Call Features

For ring tone User 1 and User2, the cadence is fixed with the on-time equals to the duration of the ring tone file and off-time equals to four seconds. The total ring duration is fixed at 60 seconds. The user ring tone names displayed on the phone LCD are derived from the ring tone file header file.

There is no need to reboot the phone after downloading a ring tone. However, if the phone is already in ring tone LCD menu while the new ring tone is downloaded, you need to exit and re-enter the ring tone menu to see the new ring tone name on the menu.

To convert a file for use as a ringtone, use the Upload Ringtone utility available from the Linksys website:

http://www.sipura.com/Documents/ringtone.zip Figure 3-5 illustrates the user interface provided by this utility.

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Configuring a Dial Plan

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Figure 3-5 Upload Ringtone Utility

For input, you can use any .wav file less than eight seconds.

Configuring a Dial Plan

The 900 Series phone allows each line to be configured with a distinct dial plan. The dial plan specifies how to interpret digit sequences dialed by the user, and how to convert those sequences into an outbound dial string.

The 900 Series phone syntax for the dial plan closely resembles the corresp onding syntax specified by MGCP and MEGACO. Some extensions are added that are useful in an end-point.

Dial Plan Digit Sequences

The plans contain a series of digit sequences, separated by the | character. The collection of sequences is enclosed in parentheses.

When a user dials a series of digits, each sequence in the dial plan is tested as a possible match. The matching sequences form a set of candidate digit sequences. As more digits are entered by the user, the set of candidates diminishes until only one or none are valid.

Any one of a set of terminating events triggers the 900 Series phone to either accept the user-dialed sequence and transmit it to initiate a call, or else to reject it as invalid. The terminating events are as follows:

• No candidate sequences remain—The number is rejected.

• Only one candidate sequence remains, and it has been matched completely—The numb er is

accepted and transmitted after any transformations indicated by the dial plan, unless the sequence is barred by the dial plan, in which case the number is rejected.

• A timeout occurs—The digit sequence is accepted and transmitted as dialed if incomplete, or

transformed as per the dial plan if complete.

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• An explicit “send” (user presses the # key)—The digit sequence is accepted and transmitted as

dialed if incomplete, or transformed as according to the dial plan if complete.

The time-out duration depends on the matching state. If no candidate sequences are as yet complete (as dialed), the Interdigit_Long_Timeout applies. If a candidate sequence is complete, but there exists one or more incomplete candidates, the <Interdigit_Short_Timeout> applies.

Table 3-5 describes the entries to use when programming the dial plan.

Table 3-5 Dial Plan Entries

Dial Plan Entry Function

*xx Allows arbitrary 2-digit star code [3469]11 Allows x11 sequences (for example, 311, 411, 611, 911) 0 Dials operator 00 Dials international operator [2-9]xxxxxx Dials US local number 1xxx[2-9]xxxxxx Dials US 1 + 10-digit long distance number

Configuring a Dial Plan

xxxxxxxxxx. Dials all other numbers, including international long distance

Dial Plan Rules

This section describes the rules that apply to configuring and interpreting dial plans.

Note White space in a dial plan script is ignored, but it may be used for readability.

Digit Sequence Syntax

Each digit sequence within the dial plan consists of a series of elements, which are individually matched to the keys pressed by the user. Elements can be one of the following:

• Individual keys 0, 1, 2 . . . 9, *, #.

• The letter x matches any one numeric digit (0 .. 9)

• A subset of keys within brackets (allows ranges): fo r example, [3 89] means 3 o r 8 or 9)

–

Numeric ranges (n-n) are allowed within the brackets: for example, [2-9] means any digit from 2 through 9)

Ranges can be combined with other keys: e.g. [235-8*] means 2 or 3 or 5 or 6 or 7 or 8 or *.

Element Repetition

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Sub-sequence Substitution

A sub-sequence of keys (possibly empty) can be automatically replaced with a different sub-sequence using an angle bracket notation: < dialed-subsequence : transmitted-subsequence >. So, for example, “<8:1650>xxxxxxx” would match “85551212” and transmit “16505551212”.

Intersequence Tones

An “outside line” dial tone can be generated within a sequence by appending a comma (,) between digits. Thus, the sequence “9, 1xxxxxxxxxx” sounds an “outside line” dial tone after the user presses 9, until the 1 is pressed.

Number Barring

A sequence can be barred (rejected) by placing a ! character at the end of the sequence. Thus, “1900xxxxxxx!” automatically rejects all 900 area code numbers from being dialed.

Chapter 3 Managing Linksys 900 Series IP Phones

Interdigit Timer Master Override

The long and short interdigit timers can be changed in the dial plan (affecting a specific line) by preceding the entire plan with the following syntax:

• Long interdigit timer: L : delay-value ,

• Short interdigit timer: S : delay-value ,

Thus, “L:8,( . . . )” sets the interdigit long timeout to 8 seconds for the line associated with this dial plan. And, “L:8,S:4,( . . . )” overrides both the long and the short time-out values.

Local Timer Overrides

The long and short timeout values can be changed for a particular sequence starting at a particular point in the sequence. The syntax for long timer override is: “L delay-value<space>“. Note the terminating space character. The specified delay-value is measured in seconds. Similarly , to change the short timer override, use: S delay-value<space>.

Pause

A sequence may require an explicit pause of some duration before continuing to dial digits, in order for the sequence to match. The syntax for this is similar to the timer override syntax: P delay-value <space>. The delay-value is measured in seconds.

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This syntax allows for the implementation of Hot-Line and Warm-Line services. To achieve this, one sequence in the plan must start with a pause, with a 0 delay for a Hot Line, and a non-zero delay for a Warm Line.

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Implicit Sequences

The 900 Series phone implicitly appends the vertical code sequences entered in the administration web server Regional parameter settings to the end of the dial plan for both Line 1 and Line 2. Likewise, if <Enable_IP_Dialing> is enabled, IP dialing is also accepted on the associated line.

Dial Plan Examples

The following dial plan accepts only US-style 1 + area-code + local-number, with no restrictions on the area code and number:

( 1 xxx xxxxxxx )

The following also allows 7-digit US-style dialing, and automatically inserts a 1 + 212 (local area code) in the transmitted number.

( 1 xxx xxxxxxx | <:1212> xxxxxxx )

For an office environment, the following plan requires a user to dial 8 as a prefix for local calls and 9 as a prefix for long distance. In either case, an “outside line” tone is played after the initial 8 or 9, and neither prefix is transmitted when initiating the call.

( <9,:> 1 xxx xxxxxxx | <8,:1212> xxxxxxx )

Configuring a Dial Plan

The following allows only placing international calls (011 call), with an arbitrary number of digits past a required 5 digit minimum, and also allows calling an international call operator (00). In addition, it lengthens the default short interdigit tim eout to 4 second s.

S:4, ( 00 | 011 xxxxx x. )

The following allows only US-style 1 + area-code + local-number, but disallows area codes and local numbers starting with 0 or 1. It also allows 411, 911, and operator calls (0).

( 0 | [49]11 | 1 [2-9]xx [2-9]xxxxxx )

The following allows US-style long distance, but blocks 9xx area codes:

( 1 [2-8]xx [2-9]xxxxxx )

The following allows arbitrary long distance dialing, but explicitly blocks the 947 area code.

( 1 947 xxxxxxx ! | 1 xxx xxxxxxx )

The following implements a hot line phone, which automatically calls 1 212 5551234.

( S0 <:12125551234> )

The following provides a warm line to a local office operator (1000) after five seconds, unless a four-digit extension is dialed by the user.

( P5 <:1000> | xxxx )

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Configuring a Dial Plan

Dial Plan Timers

The dial plan functionality is regulated by the following configurable parameters:

• Interdigit_Long_Timer

• Interdigit_Short_Timer

• Dial_Plan ([1] and [2])

Interdigit Long Timer

ParName Interdigit_Long_Timer Default 10

The <Interdigit_Long_Timer> specifies the default maximum time (in seconds) allowed between dialed digits, when no candidate digit sequence is as yet complete (see the discussion of the Dial_Plan parameter for an explanation of candidate digit sequences).

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Interdigit Short Timer

ParName Interdigit_Short_Timer Default 3

The <Interdigit_Short_Timer> specifies the default maximum time (in seconds) allowed between dialed digits, when at least one candidate digit sequence is complete as dialed (see the following discussion of Dial_Plan parameters for an explanation of candidate digit sequences).

Dial Plans

ParName Dial_Plan[n] for Each Line n

Default

The Dial_Plan parameters contain the actual dial plan scripts for each line n, where n is a number from 1 to 4.

(*xx | [3469]11 | 0 | 00 | <:1408>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011x. )

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System Administration

This section describes features and operations required for administering and maintaining the basic function of the SPA9 00 Series IP ph one. It includes the following topics:

• Reboot and Restart, page 3-28

• Factory Reset, page 3-28

• Password Protection, page 3-28

• Managing the Time/Date, page 3-28

• Daylight Saving Time, page 3-29

• Using Star Codes to Activate/Deactivate Service s, page 3-3 0

• Disabling Services, page 3-31

• Error and Log Reporting, page 3-32

Reboot and Restart

System Administration

Reboot cold boots the phone and restart DHCP. Restart warm boots the phone, without restarting DHCP. Both options require the user to confirm before proceeding.

Factory Reset

Factory reset resets all parameters to default value. The personal directory and call logs are also cleared. This option require the user to confirm before proceeding.

Password Protection

The user can configure a password to the phone to protect against access to the certain options.

Managing the Time/Date

The 900 Series phone obtains the current time information in one of the following ways:

• NTP Server—One or two NTP servers can be configured to the phone. When it first boots up, the

phone tries to contact the NTP server to get the current time. Then the phone periodically synchronizes the current time with the NTP server. The synchronization period is fixed at 1 hour. In between the update, the phone tracks the time with its own internal clock.

• SIP Messages—Each SIP message (request or response) sent to the SPA may contain a Date header

with the current time information. If the header is present, the SPA uses it to update its current time.

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• Manual Setup—900 Series phone also lets the user manually enter the current time and date from

the phone GUI or from the web page. Howeve r, this value is overridden by the NTP t ime or SIP Message Date whenever they are presented to the phone. Manual setup requires the user to enter the time in 24 hour format only.

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System Administration

The time served by the NTP Server and the SIP Date Header are expressed in GMT time. The local time is obtained by offsetting the GMT according to the time zone of the region. The <Time Zone> parameter can be configured from the web page or through provisioning. This time can be further offset using the <Time Of fset (HH/mm)> parameter , which must be entered in 24-hour format. This parameter can also be configured from the LCD display. The <Time Zone> and <Time Offset (HH/mm)> offset values are not applied to the manual time/date setup.

Daylight Saving Time

The 900 Series phone supports auto adjustment for daylight saving time. To enable this feature, the administrator must configure the <Daylight Saving T ime Rule> parameter on the Regional tab. This parameter is a rule with three fields separated by semicolon (;) as shown below:

start=<start-time>;end=<end-time>;save=<save-time>

Where:

• <start-time> and <end-time> are of the form: <month>/<day>/<weekday>[/HH:[mm[:ss]]]

• <save-time> is of form: [/[+|-]HH:[mm[:ss]]]

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• <month> = 1,2,12 (for Jan, Feb, Dec)

• <day> = [+|-]1,2,3,...,,31

• <weekday> = 1, 2, 3,..., 7 (for Mon, Tue, ..., Sun), or 0 which has special meaning

• HH = hour (0-23)

• mm = minute (0-59)

• ss = second (0-59)

<start-time> and <end-time> specify the start and end time and date of daylight saving time and <save-time> is the amount of hour/min/sec to add to the current time during daylight saving period.

The <save-time> value can be preceded by a negative (-) sign if subtraction is desired instead of addition. If <weekday> is 0, it means the date to start or end daylight saving is at exactly the given date. In that case, the <day> value must not be negative. If <weekday> is not zero, then the daylight saving starts or ends on the <weekday> on or after the given date if <day> is positive, or on or before the given date if <day> is negative. If <day> is 1, it means the <weekday> on or before the end-of-the-month (in other words the last occurrence of < weekday> in that month).

Optional values inside [ ] are assumed to be 0 if they are not specified. Midnight means 0:0:0 of the given date.

The following example configures daylight savings time for the U.S, adding one hour starting at midnight on the first Sunday in April and ending at midnight on the last Sunday of October:

start=4/1/7/0:0:0;end=10/31/7/0:0:0;save=1 start=4/1/7;end=10/-1/7;save=1 start=4/1/7/0;end=10/-1/7/0;save=1

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The following example configures daylight savings time for Egypt, starting at midnight on the last Sunday in April and ending at midnight on the last Sunday of September:

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start=4/-1/5;end=9/-1/4;save=1 (Egypt)

The following example configures daylight savings time for New Zealand, starting at midnight on the first Sunday of October and ending at midnight on the third Sunday of March.

start=10/1/7;3/22/7;save=1 (New Zealand)

Using Star Codes to Activate/Deactivate Services

The 900 Series phone accepts star codes to activate or deactivate certain services on the phone. Below is a list of services that are accessible with a star code, with the default star code shown in parenthesis:

• Call Return (*69)—Calls the last caller, regardless which extens ion.

• Call Back(*66)—Periodically r edials the l ast busy number (every 30s by default) until it rings or

until the trial expires (30 min by default), regardless which extension. Only 1 call back operation can be ordered at a time. A new order automatically cancels the previous order.

• Cancel Call Back(*86)—Cancels the last call back operation.

System Administration

• Call Forward All (*72)—Call f orwards all inbound c alls. Appl ies to pr imary exte nsion only

• Cancel Call Forward All (*73)—Cancels call forward all. Applies to primary extension only

• Call Forward Busy (*90)—Call forwards on busy. Applies to primary extension only.

• Cancel Call Forward Busy (*91)—Cancels call forward on busy. Applies to primary extension only.

• Call Forward No Answer (*92)—Call forwards if no answer. Applies to primary extension only.

• Cancel Call Forward No Answer (*93)—Cancels call forward no answer. Applies to primary

extension only.

• Block Caller ID Always (*67)—Blocks caller ID on all outbound calls. Applies to all extensions.

• Block Caller ID Per Call (*81)—Blocks caller ID on the next outbound call (on the current call

appearance only).

• Unblock Caller ID Always (*68)—Unblocks caller ID on all outbound calls. Applies to all

extensions.

• Unblock Caller ID Per Call (*82)—Unblocks caller ID on the next ou tbound call (on the current call

appearance only).

• Secure All Calls (*16)—Defaults to prefer to use encrypted media for all out bound calls. Applies t o

all extensions.

• Secure No Calls (*17)—Defaults to prefer to use unencrypted media for all outbound calls. Applies

to all extensions.

• Secure Next Call (*18)—Prefers to use encrypted media for the next outbound call (on this call

appearance only).

• Do Not Secure Next Call (*19)—Prefers to use unencrypted media for the next outbound call (on

this call appearance only).

• Do Not Disturb (*78)—Applies to all extensions.

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• Cancel Do Not Disturb (*79)—Applies to all extensions.

• Block Anonymous Calls—Applies to all extensions.

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Chapter 3 Managing Linksys 900 Series IP Phones

• Cancel Block Anonymous Calls—Applies to all extensions.

• Blind Transfer (*98)

• Prefer G.711u (*017110)—Sets preferred codec to G.711u for next outbound call.

• Prefer G.711a (*017111)—Sets preferred codec to G.711a for next outbound call.

• Prefer G.723 (*01723)—Sets preferred codec to G.723 for next outbound call.

• Prefer G.726r16 (*0172616)—Sets preferred codec to G.726r16 for next outbound call.

• Prefer G.726r24 (*0172624—Sets preferred codec to G.726r24 for next outbound call.

• Prefer G.726r32 (*0172632)—Sets preferred codec to G.726r32 for next outbound call.

• Prefer G.726r40 (*0172640)—Sets preferred codec to G.726r40 for next outbound call.

• Prefer G.729a (*01729)—Sets preferred codec to G.729a for next outbound call.

• Force G.711u (*027110)—Forces G.711u for next outbound call.

• Force G.711a (*027111)—Forces G.711a for next outbound call.

• Force G.723 (*02723)—Forces G.723 for next outbound call.

• Force G.726r16 (*0272616)—Forces G.726r16 for next outbound call.

• Force G.726r24 (*0272624—Forces G.726r24 for next outbound call.

• Force G.726r32 (*0272632)—Forces G.726r32 for next outbound call.

• Force G.726r40 (*0272640)—Forces G.726r40 for next outbound call.

• Force G.729a (*02729)—Forces G.729a for next outbound call.

When off-hook dialing, the star code is processed by the phone as soon as it is entered and processed by the phone. When on-hook dialing, the user can pre-dial multiple star codes before the target number. The star codes are handled in the order they are entered, before the target number is dialed. The *code soft key indicates the *codes available. Star codes can also be included in a directory entry or redial list.

In addition there are two special codes:

• Referral Services Codes

This is a list of star codes separated by a vertical bar (|), such as *79|*199|*09, wher e enter ing each member triggers a blind transfer operation. The transfer target number in this case is prepended by the corresponding star code, whereas the blind transfer itself contains the target number only without the star code. Like a normal blind transfer , a referral service code can only be entered when the last call is placed on hold.

• Feature Dial Services Codes

This is a list of star codes separated by a vertical bar (|), where entering each member causes the next outbound call target number to be prepended by the corresponding star code.

Disabling Services

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Services handled locally by the phone can be disabled by the administrator in one or two ways:

• Directly disabling the service in the configurable profile,

• Emptying out the *code associated with the service.

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Services that can be disabled (or enabled) by the first method include the following:

• Block Caller ID (including blocking and unblocking, per call and always)

• Block Anonymous Calls

• Do Not Disturb

• Secure Call (including enabling and disabling, per call and always)

• Call Forward All

• Call Forward Busy

• Call Forward No Answer

• Call Back

• Conference Call (no star code)

• Attended Call Transfer (no star code)

• Blind Call Transfer

Services that can be disabled (or enabled) by the second method include any service that has a star code assignment in the 900 Series phone configuration profile. Note that if a service is enabled, but the corresponding *code is emptied, the service can still be enabled/disabled by the end- user from the phone LCD or the administration web server.

System Administration

If a service is disabled, any soft key associated with that service is hidden on the LCD. These are: dnd), -dnd), cfwd) and -cfwd). Any menu item associated with a disabled service is preceded with an exclamation mart (!) and no soft keys is visible to select that entry.

Error and Log Reporting

The SPA Error Status Code (ESC) is used to indicate the current operation status of the SPA unit. An error state can be a relatively long transient state or a steady state.

The Error Status Code is a 4 digit number. The first digit indicates the error class: 1xxx represents normal operation states while 2xxx— 9xxx represent error states that must be fixed for the unit to function properly. The s tatus code values can be read from the IV R option XXX or from the SPA web-page.

LED Blink Rate Definitions

• ON—LED remains solid on

• OFF—LED remains solid off

• LONG (Long On)—3.0 second on, 1 second off continuously

• FAST—0.125 second on, 0.125 second off continuously

• SLOW—0.5 second on, 0.5 second off continuously

• VSLO (Very Slow)—1.0 second on, 1.0 second off continuously

• HB (Heart Beat)—0.1 second on, 0.1 second off, 0.1 second on, 1 second off continuously

Note The Link LED flashes on transmit and receive (TX/RX) of pa ckets. The LED displays solid off i f no link

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is available. The LED displays solid on if link is up but no TX/RX activity is present.

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Troubleshooting FAQ

This section describes some common problems and solutions.If the suggestions provided here do not solve the problem, contact the appropriate technical support department.

If you are an end user of LVS products and need technical support, contact the reseller or Internet telephony service provider (ITSP) that supplied the equipment.

Technical support contact information for authorized Linksys Voice System partners is as follows:

• LVS Phone Support (requires an authorized partne r PIN)

888 333-0244 Hours: 4am-6pm PST, 7 days a week

• E-mail support

voipsupport@linksys.com

When you contact the VoIP network administrator or service provider, have the model, serial number, and telephone number ready for technical support. The serial number is printed on the bottom of the phone.

Chapter 3 Managing Linksys 900 Series IP Phones

Why do the line status lights on my SPA900 Series occasionally flash red?

A flashing red line status light on the SPA900 Series phone indicates a l ocal LAN issue. If statistics on the switch show a high number of collisions on the port s connect ed to the 90 0 Seri es phone, y ou can reduce the speed on the ports to 10mb half duplex to eliminate the collision problems.

Why can I not leave a voicemail message longer than 30 seconds?

If you are using 1.0.0.3 firmware, upgrade to version 1.0.0.5.

Why does the phone does not turn on or boot?

Perform the following steps:

1. Check all the connections into the phone (5 Volt Power supply, RJ 45 Ethernet connection, and

Handset cord.)

2. Check the wall socket and confirm that AC power is available.

3. Try plugging another electrical device into the socket to confirm power.

Why does the phone not make or receive calls?

If the phone does not make or receive calls, perform the following steps.

1. Press the Menu button, then 11, then 2.

This provides the registration information for Extension 1 of the phone. The screen should display E1 Registered and other information in the format provided below:

Text Box: E1 Registered Last Reg: 08:33 1/3 Next Reg in: 13s Status cancel cancel

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There may be a problem if the screen displays Not Registered.

2. Write down whether the phone is Registered or Not Registered.

3. To exit from this option, press the Menu button.

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4. Get the current IP address of the phon e by pressing Menu, 9, 2.

The phone displays the current IP address.

5. If there is no address or the address does not appear correct for your network, restart the phone, the

SPA9000, and the router or DHCP server.

Why are voice calls of poor sound quality?

The source of this issue is often one of the following two issues:

• The phone or call controller may have been configured improperly.

• The transport network has performance issues.

To troubleshoot this problem, gather the following information: Identify when and where the issue occurred. Look at the Redial List (calls that the user made) by

pressing Menu, 3, 1, and write down the calls that were affected. Then look at the Answered Calls by pressing Menu, 3, 2, and write down the calls that were affected.

Identify the type of voice quality issue (for example, part of the conversation is dropped, excessive echo, unclear or garbled speech, one way conversation, or low volume)

6. Get the current IP address of the phon e by pressing Menu, 9, 2.

7. Contact the network administrator or service provider. Be prepared to provide the call log

information, the voice quality issue type, the current IP ad dress an d registration status (Registered, Not Registered).

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CHAPTER

LCD Command Reference Guide

This chapter provides a systematic reference for the config uration and monitoring options available fro m the LCD display of SPA900 Series IP phones. All models provide an LCD display except for the SPA901. The SPA901 uses the Interactive Voice Response (IVR) interface, described in the

Interactive Voice Response Interface” section on page 2-16. This chapter includes the following

sections:

• 1 Directory, page 4-2

• 2 Speed Dial, page 4-3

• 3 Call History, page 4-3

• 4 Ring Tone, page 4-4

• 5 Preferences, page 4-5

• 6 Call Forward, page 4-7

• 7 Time/Date, page 4-8

• 8 Voice Mail, page 4-8

• 9 Network, page 4-8

“Using the

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• 10 Product Info, page 4-12

• 11 Status, page 4-13

• 12 Reboot, page 4-14

• 13 Restart, page 4-14

• 14 Factory Reset, page 4-15

• 15 Set Password, page 4-15

• 16 Set LCD Contrast, page 4-15

• 17 CallPark Status, page 4-15

• 18 Language (SPA922, 942, and 962), page 4-16

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1 Directory

The directory stores the list of the phone numbers and names. This section describes how to enter and save phone numbers and names into the directory. It includes the following sections:

• Entering Names and Numbers into the Directory, page 4-2

• Entering Directory Names, Numbers and Ring Default, page 4-2

Entering Names and Numbers into the Directory

Shortcut: Menu, 1, add, name, scroll down, number, scroll down or save

To enter the Directory option, press the Menu button and then press the 1 button. Alternatively, you may press the Menu button and the Directory option is highli ghted. Then press the select soft key . Aft er pressing select, the menu displays New Entry and existing entries in the directory below. To add a New Entry, press the add button, (left soft key). To cancel, press the cancel soft key.

After pressing the add button, the menu shows the Name, Number, and Ring Default with the cursor flashing in the first character of the Name area.

Chapter 4 LCD Command Reference Guide

Entering Directory Names, Numbers and Ring Default

After pressing add, the user is prompted to enter a Name. To enter a letter of a name, press t he letter on the numeric keypad.

For example, press 2, and the letter a appears. If the cursor is in the first letter of name, it is a capital letter. Every two seconds, the cursor flashes. If the 2 is NOT pressed again before the cursor flashes, the phone accepts the A as the entry and it moves to the next position. To obtain a different letter/number associated with the 2 (a, b, c, 2, A, B, C), press 2 again before the cursor flashes. The letters/numbers are displayed at the bottom of the screen and the letter/number that is entered is highlighted.

The first letter in a directory name defaults to a capital letter and subsequent letters in the name defaults to lower case. After entering the last letter, scroll d own. The cursor is moved to the Number field. Enter the telephone number associated with this name. When you are finished entering the number, scroll down. The cursor is moved to the Ring field. Enter the Ring type for this entry, and then press ok. The entry is saved in the directory.

Press the delChr soft key to correct a mistake. Each time the delChr soft key is pressed one letter from the entered string is removed from the display.

To enter a phone number into the directory, simply dial the number including the 1 and the Area Code (for example, 14085721111).

The same screen can be used to set the phone to a distinctive ring for each caller. To set the ring, move the cursor to the Ring Default area by pressing the down arrow button after entering the phone number instead of pressing save per above. Use the arrow button to scroll to and highlight the ring f or this person and press the select soft key. When the entry is complete, press the save button (left soft key). After the information has been saved, the phone displays the directory with the name and number.

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Note Be sure to press the save button after entering information. Otherwise, the information is not saved and

will not appear in the directory.

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2 Speed Dial

The Speed Dial option allows a single number to be used to automatically dial a telephone, IP address, or URL from the directory. This section provides directions for setting up and using speed dial numbers.

T o set-up a speed dial entry , press the Menu button and then press 2. Alternatively , press the Menu button and scroll down until SpeedDial is highlighted and then press the select soft key.

The Speed dial menu indicates numbers available for assignment as <Not Assigned>.To assign an entry to an available number, press th e number (or scroll down un til it is hi ghl ighted), th en press the edit soft key. The menu displays Enter Speed Dial 1. After pressing edit, the menu displays Enter Speed Dial Number 1. To enter a number or a letter, press the appropriate number.There are two options to create a speed dial entry:

• Enter a telephone number, an IP ad dress or a URI into a speed dial memory location. When finished,

press ok (left soft key). The entry is added to speed dial.

• Enter a name and number from the directory. As the user enters in letters, the phone displays names

in the directory that match the letters displayed. Use the scroll butt ons at any time to highlight to the name or type the name in completely. After highlighting a name (or typing in the name), press ok. The entry is added to speed dial.

Press the delChr soft key if a mistake is entered. Pressing the delChr soft key removes one letter from the entered string at a time.

There are two ways to use speed dial entries:

• Dial the speed dial number and then lift the handset.

• Lift the handset, dial the speed dial number, and then #.

2 Speed Dial

3 Call History

To access Call History, press the Menu button and the press 3. Alternatively, the user ca n press Menu and then can scroll down until call history is highlighted and then press the select soft key. The shortcut is Menu, 3.

The Call History option provides three options:

• Redial List

• Answered Calls

• Missed Calls

To select an option, press the corresponding number or scroll down until the appropriate option is highlighted and press the select soft key. To return to the previous menu use the cancel option (righ t soft key).

Redial List

Shortcut: Menu, 3, 1

After selecting this option, the phone displays the call entries in the Redial List. The details of the call are displayed including the name if available, along with the telephone number,

the date, and the time of the call. To dial an entry, scroll through the list to highlight the number and press the dial soft key. To return to the Menu option, press the cancel soft key.

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4 Ring Tone

Answered Calls

Shortcut: Menu, 3, 2

After selecting this option, the phone displays the call entries in the Answered Call List. The details of the call are displayed, including the name if available, along with the telephone number,

the date, and the time of the call. To dial an entry, scroll through the list to highlight the number and press the dial soft key. To return to the Menu option, press the cancel soft key..

Missed Calls

Shortcut: Menu, 3, 3

After selecting this option, the phone displays the call entries in the Missed Calls List. The detail s of the call are displayed, including the name if available, along with the telephone number, the date, and the time of the call. To dial an entry, scroll through the list to highlight the number and press the dial soft key. To return to the Menu option, press the cancel soft key..

Chapter 4 LCD Command Reference Guide

4 Ring Tone

Shortcut: Menu, 4, select extension, change, select ring tone

This option lets the user select a different ring tone for each extension that th e phone supports. It l ets the user personalize his phone and can be useful in a shared office environment to provide each user with have a distinctive ring tone.

T o set or change the ring tone for a specific extension, press Menu and then select ring to nes by pressing

4. Alternatively, press the Menu button and scroll d own to the ring to ne option and press select using the left soft key. To exit this option and return to the previous menu, press the cancel soft key.

When the Ring Tone option is selected, the ring tone used for each extension is displayed. To change a Ring tone, either enter the corresponding number or scroll down until the selection is highlighted and press the change soft key. To return to the previous menu, press the cancel soft key.

T o select a ring tone , press the corresponding num ber or highlight the opti on. To listen to the ring tones, press the play soft key. To pick the ring tone desired for this extension, press the select soft key. After pressing select, the display returns to the menu showing the extensions and the ring tones associated with them.

5 Preferences

This option allows you to set the preferences on the phone including the following:

• 5.1 Block Caller ID

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• 5.2 Block Anonymous callers

• 5.3 Do Not Disturb

• 5.4 Secure Call

• 5.5 Dial Assistance

• 5.6 Preferred Audio Device

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T o select the Preferences option, press the Menu button and then press 5 or scroll down to the Preference option, highlight it, and press the select soft key. To select an option from the Preferences menu, press the number of the selection or scroll dow n to the se lection a nd press th e select soft key. To enable or disable an option, toggle between Yes and No by pressing the y/n soft key.

5.1 Block Caller ID

Shortcut: Menu, 5, edit, y/n, ok

When Block Caller ID is enabled, the calling number does not appear to the called party. To select Block Caller ID, press th e Menu button and th en pre ss 5 for preferences, or scroll down until

Preferences is highlighted, and press the select soft key. With Block Caller ID No highlighted, press the edit soft key.

Enable or disable the option by pressing the y/n soft key and press ok. The display returns to the Preferences Menu.

5.2 Block Anonymous Call

Shortcut: Menu, 5, select, 2, edit, y/n, ok

When this feature is enabled, calls from callers that do not present their Caller ID are not received. To select Block Anonymous Calls, press the Menu button and then press 5, or scroll down until the

selection is highlighted and press the select soft key. Press 2 or scroll down until Block Anonymous Calls is highlighted and then press the edit soft key.

Enable or disable the option by pressing the y/n soft key and press ok. The display returns to the Preferences Menu.

5 Preferences

5.3 Do Not Disturb

5.4 Secure Call

Shortcut: Menu, 5, 3, edit, y/n, ok

Do Not Disturb routes the call to voicemail or presents a busy signal if voicemail is not available. From the Preferences menu, press 3 or scroll down until the selection is highlighted and then press the

edit soft key. This option is on the second preferences screen so scroll past the Block Caller ID, and Block Anonymous caller options to get to the Do Not Disturb option.

Enable or disable the option by pressing the y/n soft key and press ok. The display returns to the Preferences Menu.

Shortcut: Menu, 5, 4, edit, y/n, ok

When a call is secure, the voice conversation is encrypted so that others cannot eavesdrop on the conversation. To enable this feature the 900 Series phone must have a mini-certificate installed.

From the Preferences menu, press 4 or scroll down until the selection is highlighted and then press edit. This option is on the second Preference screen; scroll past the first three preferences.

Enable or disable the option by pressing the y/n soft key and press ok. The display returns to the Preferences Menu.

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5.5 Dial Assistance

Shortcut: Menu, 5, 5, edit, y/n, ok

The Dial Assistance option allows the 900 Series phone to automatically suggest a n umber on the display based on entries in the directory or call logs. Use the down arrow button to select the number to be dialed. Press dial to complete the call

Enable or disable the option by pressing the y/n soft key and press ok. The display returns to the Preferences Menu.

5.6 Preferred Audio Device

Shortcut: Menu, 5, 6, edit, y/n, ok

From the Preferences menu, press 6 for Preferred Audio Device or scroll down until the selection is highlighted and then press edit.

Use the option soft key to toggle between the two options, Speaker or Handset. W ith the Preferred Audio Device showing the desired selection, press ok. The display returns to the Preferences menu.

Chapter 4 LCD Command Reference Guide

6 Call Forward

To view or change the call forwarding setting, press the Menu button and press the 6 button or use the arrow button to scroll down to Call Forward and press the edit soft key. The Call Forward menu provides four call forwarding options:

• 6.1 CFWD All Number—Forwards all calls to a single number

• 6.2 CFWD Busy Number—Forwards calls to another number when the phone is busy

• 6.3 CFWD No Ans Number—Forwards calls to another number if there's no answer

• 6.4 CFWD No Ans Delay—Forwards calls if there is no answer after a time delay

6.1 CFWD All Number

Shortcut: Menu, 6, edit, phone_number, Ok

This option forwards all calls to the specified number. Enter the phone number for calls to be forwarded using the same method as dialing a number. For an

external number, include the 1 and the area code (for example, 14085551212). Erase mistaken numbers, one at a time using the delChr soft key. After the number is entered correctly, press ok.

The forwarding number is displayed on the Call Forward option. Calls are not forwarded if a telephone number is not entered followed by ok.

6.2 CFWD Busy Number

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Shortcut: Menu, 6, 2, edit, phone_number, ok

This option forwards calls to another number when the phone is busy.

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To forward all calls to specific telephone number when the phone is busy Press Menu, Press 6, press 2 or scroll down until CFWD Busy Number is highlighted and press the edit soft key . Enter the telephone number to which to forward calls when the phone is busy. After the number is entered correctly, press ok. The forwarding number is displayed on the Call Forward sub menu. Calls are not forwarded unless a telephone number is entered.

6.3 CFWD No Ans Number

Shortcut: Menu, 6, 3, edit, phone_number, ok

This option forwards calls to another number if there is no answer. Press the edit soft key and enter the phone number to which calls should be forwarded when the phone

is not answered. After the number is entered correctly, press ok. The forwarding number is displayed on the Call Forward sub menu. Calls are not forwarded if a telephone number is not entered followed by ok.

6.4 CFWD No Ans Delay

Shortcut: Menu, 6, 4, edit, phone_number, ok

This option sets the time interval after which forwarding occurs. Press the edit soft key and the number of seconds to elapse before the phone forwards the call. Calls are

forwarded to the telephone number entered using the CFWD No Ans option after the specified time has elapsed.

7 Time/Date

Shortcut: Menu, 7 ,date, time, offset, save

To set the date and time, press Menu and then press 7, or scroll down until the selection is highlighted and then press select. Enter the date and time in the following format:

• Date—m/d/y

• Time—H:m:s

• Offset—+/-H:m:s

To enter a / or : use the * (star) key. For offset value, use the * (star) key for + and the # (pound) key for

-. After entering the required values, press the save soft key, to save the information into the phone memory. Erase mistaken numbers, one at a time using the delChr soft key.

8 Voice Mail

Shortcut: Menu, 8, voicemail_number, save

To set the number to call voicemail, press Menu and then press 8, or scroll down until the selection is highlighted and press the select soft key.

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Chapter 4 LCD Command Reference Guide

When the display prompts for the Voice Mail Number, enter the number as if dialing the number. Erase mistaken numbers, one at a time using the delChr soft key. Then press the save button to save the number. After the number is saved, use the mailbox key on the right hand side of the phone (this is the third button up from the Menu button) to dial the mailbox.

To view or modify network parameters, press the Menu button and then press 9, or scroll down until Network is highlighted and then press the select soft key. This menu di splays the IP ne twork configuration settings.

The following options are available from the Network menu:

• 9.1 DHCP

• 9.2 Current IP Address

• 9.3 Host Name

• 9.4 Domain

• 9.5 Current Subnet Mask

• 9.6 Current Gateway

• 9.7 Enable Web Server

• 9.8 Non DHCP IP Address

• 9.9 Non DHCP Subnet Mask

• 9.10 Non DHCP Default Route

• 9.11 Non DHCP DNS 1

• 9.12 Non DHCP DNS 2

• 9.13 Non DHCP NTP Server 1

• 9.14 Non DHCP NTP Server 2

9.1 DCHP

Shortcut: Menu, 9, edit, y/n, ok

When DHCP is enabled, the phone obtains its IP address from a DHCP server. To view DHCP status, press the Menu button and press 9, or scroll down to the Network option and

press the select soft key. The displays shows 1 DHCP Enable with a Yes or No to indicate the current status.

To enable or disable DHCP, press the edit soft key and press the y/n soft key to toggle between Yes and No. After making the selection, press the ok button and the display returns to the Network menu.

9.2 Current IP Address

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Shortcut: Menu, 9, 2 save

To view the current IP address, press 2 on the Network menu, or scroll down until Current IP Address is highlighted and press the select soft key. The Current IP Address is set in one of the following ways:

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• Receive the address from a DHCP server by enabling DHCP (Option 9.1)

• Manually set the address using the Non DHCP IP Address option (Option 9.8)

9.3 Host Name

Shortcut: Menu, 9, 3 save

The Host Name is set by the adm inistrator. The user cannot change the setting.To view the Host Name, press 3 from the Network menu, or scroll down to Host Name and press the select soft key.

9.4 Domain

Shortcut: Menu, 9, 4 save

The Domain Name displays the name of the domain in which the phone is operating and is not user configurable. To view the Domain, press 4 from the Network menu, or scr oll down to highlight Domain and press select.

9.5 Current NetMask

9 Network

Shortcut: Menu, 9, 5 save

T o view the Current NetMask, press 5 from the Network menu, or scroll down to highlight NetMask and press the select soft key. The Current NetMask is set in one of the fo llowin g ways:

• Receive the address from a DHCP server by enabling DHCP (option 9.1)

• Manually set the address using the Non DHCP Subnet Mask option (Option 9.9)

9.6 Current Gateway

Shortcut: Menu, 9, 6 save

To view the Current Gateway, press 6 on the Network menu, or scroll down to highlight Current Gateway and press the select soft key.

The Current Gateway is set in one of the following ways:

• Receive the address from a DHCP server (the user cannot edit it)

• Manually set the address using the Non DHCP Default Route option (Option 9.10)

9.7 Enable Web Server

Shortcut: Menu, 9, 7, y/n, ok

To view the status of the Web Server, press 7 from the Network menu, or scroll down to highlight Enable Web Server and press the select soft key. To enable or disable the Web Server, press the edit soft key. Press the y/n soft key to toggle between Yes to No and press ok.

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9.8 Non DHCP IP Address

Shortcut: Menu, 9, 8, ip_address, ok

T o manually set the IP add ress, press 8 from the Network menu, or scrol l down to hig hlight Non DHCP IP Address and press the select soft key. Then press the edit soft key. The display shows the option to

enter an IP address. enter the IP address here. This IP address is always used by the phone (it is a static IP address). After entering IP address, press ok.

9.9 Non DHCP Subnet Mask

Shortcut: Menu, 9, 9, subnet_mask, ok

To manually set the subnet mask, press 9 from the Network menu, or scroll down to highlight Non DHCP Subnet Mask and press the select soft key . Then press the edit soft key . After entering the su bnet mask, press ok.

9.10 Non DHCP Default Route

Shortcut: Menu, 9, 10, IP ADDRESS, ok

T o edit the default gateway (router), press 10 or scroll down to highlight Non DHCP Defa ult Route and press the select soft key. Then press the edit soft key. After entering the default route, press ok..

Chapter 4 LCD Command Reference Guide

9.11 Non DHCP DNS 1

Shortcut: Menu, 9, 11, ip_address, ok

T o edit the pri mary DNS server used by the phone, press 11 from the Network menu, or scroll down and highlight Non DHCP DNS 1 and press the select soft key. Then press the edit soft key. After entering the IP address of the server, press ok.

9.12 Non DHCP DNS 2

Shortcut: Menu, 9, 12, ip_address, ok

To edit the secondary DNS server used by the phone, press 12 from the Network menu, or scroll down and highlight Non DHCP DNS 1 and press the select soft key. Then press the edit soft key. After entering the IP address of the server, press ok.

9.13 Non DHCP NTP Server 1

Shortcut: Menu, 9, 13, ip_address, ok

The Network Time Protocol (NTP) server sets the time used by the IP phone. To edit the primary NTP server used by the phone, press 13 from the Network menu, or scroll down and highlight Non DHCP DNS 1 and press the select soft key. Then press the edit soft key. After entering the IP address of the server, press ok.

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To change the phone to use a Non DHCP NTP Server 1, press the edit soft key. The display shows the option to enter an IP address. enter the IP Address of the NTP Server1. This IP address is always used by the phone, it is a static IP address. Erase mistaken numbers, one at a time u s ing the delChr soft key. After IP address has been entered, press ok. The display returns to the Non DHCP NTP Server1 setting. The phone uses the IP address immediately.

9.14 Non DHCP NTP Server 2

Shortcut: Menu, 9, 14, ip_address, ok

To edit the secondary NTP server used by the phone, press 14 from the Network menu, or scroll down and highlight Non DHCP DNS 1 and press the select soft key. Then press the edit soft key. After entering the IP address of the server, press ok.

10 Product Info

To view the parameters in the Product Info, press the Menu button and then press 10, or scroll down until Product Info is highlighted and then press the select soft key. The information displayed cannot be edited. The options on the Product Info menu are as follows:

• 10.1 Product Name

10 Product Info

• 10.2 Serial Number

• 10.3 Software Version

• 10.4 Hardware Version

• 10.5 MAC Address

• 10.5 Client Cert

10.1 Product Name

Shortcut: Menu, 10, select

To view the Product Name, press the Menu button and press 10, or scroll down to the Product Info menu, and press the select soft key. The Product Name is also printed on the bottom of the phone.

10.2 Serial Number

Shortcut: Menu, 10, 2, select

To view the Serial Number, press 2 from the Product Info menu, or scroll down to highlight Serial Number and press the select soft key. The serial number is also printed on the bottom of the phone.

10.3 Software Version

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Shortcut: Menu, 10, 3, select

To view the software version, press 3 from the Product Info menu or scroll down to highlight Software Version and press the select soft key.

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10.4 Hardware Version

Shortcut: Menu, 10, 4, select

T o view the hardware version, press 4 from the Product Info menu or scroll down to highlight Hardware Version and press the select soft key.

10.5 MAC Address

Shortcut: Menu, 10, 5, select

To view the MAC address, press 5 from the Product Info menu or scroll down to highlight MAC Address and press the select soft key. The MAC address is also printed on the bottom of the phone.

10.6 Client Cert

Shortcut: Menu, 10, 6, select

The phone can be configured for secure provisioning using the factory-installed security certificate. To determine if the Client Certificate is installed on the phone, press 6 from the Product Info menu, or scroll down to the highlight Client Cert, and press the select soft key.

Chapter 4 LCD Command Reference Guide

11 Status

Phone

To view the status of the phone and extension parameters, press the Menu but ton and press 11, or scroll down until Status is highlighted and then press the select soft key. This menu displays read-only information about the status of the phone and provides the following options:

• Phone

• EXT 1

• EXT 2

• EXT 3

• EXT 4

• Line 1

• Line 2

• Line 3

• Line 4

Shortcut: Menu, 11, select

T o view the phone status, press the Menu button and then 11 or scroll down to Status and press the select soft key. The display provides the time and date, Up time, Transmitted packets (TX) and Received Packets (RX).

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Ext 1/2/3/4

Shortcut: Menu, 11, 2 or 3 or 4 or 5 select

T o see the settings for an extension, from the Status menu, press 2 for Ext 1, 3 for Ext 2, 4 for Ext 3, and 5 for Ext 4.

The LCD displays Registered if the extension is registered, the last registration date and time, and the time in seconds until the next registration. If the extension is not registered, the phone displays Not Registered.

Line 1, 2,3,4

Shortcut: Menu, 11, 6 or 7 or 8 or 9 select

To see the settings for a line, from the Status menu, press 6 for Ext 1, 7 for Ext 2, 8 for Ext 3, and 9 for Ext 4. The display shows the status of the line including if the line is ready and what tone is used.

12 Reboot

Shortcut: Menu, 12, ok This option lets the user reboot the phone. T o select Reboot, press the Menu button and press 12, or scroll

down until the selection is highlighted and then press select soft key. The LCD displays Confirm System Reboot ?. The display shows two options Reboot or Cancel. Select Reboot to reboot the phone. The option is only displayed for 5 seconds. Aft er 5 seconds, the display retu rns to the Me nu option.

13 Restart

Shortcut: Menu, 13, ok This option lets the user restart the phone. T o select Restart , press the Menu button and press 13, or scroll

down until the selection is highlighted and then press the select soft key. The menu displays Confirm System Restart? This display shows the options ok o r cancel. Sel ect ok to

restart the phone. This option is only displayed for 5 seconds. After 5 seconds, the display is returned to the Menu option.

14 Factory Reset

Shortcut: Menu, 14, ok

This option lets the user reset the parameters to the factory settings. To select Factory Reset, press the Menu button and press 14, or scroll down until the selection is highlighted and press the select soft key.

The menu displays All parameters are reset to factory default values, continue? The display shows two options: ok or cancel. Select ok to change the settings of the phone to the factory default settings. This option is only displayed for 5 seconds. After 5 seconds, the display is returned to the Menu option.

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15 Set Password

Shortcut: Menu, 15, password, ok

This option lets the user set a password to protect th e phone from unauthorized use. To select this option, press the Menu button and press 15, or scroll down until the selection is highlight ed and press th e select soft key.

When the LCD displays New Password, enter the password and press the save soft key.

Caution If the password is lost, to recover the phone you must perform a factory reset. A factory reset erases all

current settings, including the directory and call logs.

16 Set LCD Contrast

Shortcut: menu, 16, scroll knob left or right, save

This option lets the user adjust the LCD contrast. To use this option, press the Menu button and press 16 scroll down to highlight this option and press the select soft key.

Press the Navigation button right or left t o ad just the LC D c ont rast up or dow n. Pre ss the save soft key when finished, or the cancel soft key to keep the previously selected LCD contrast.

Chapter 4 LCD Command Reference Guide

17 CallPark Status

Shortcut: menu, 17

This option lets the user display the status of the Ca llPark feat ure. When this option is selected, the LCD displays the Requesting CallPark Status messa ge.

The LCD then displays the CallPark Status or displays the Request CallPark Status Failed message and the request is retried.

To cancel the request press Cancel.

18 Language (SPA922, 942, and 962)

Shortcut: menu, 18, Edit, Option, OK, Save

This option is available only with Release 5.1.4 or later of the SPA922, SPA942, and SPA962. This option lets the user change the language of the phone from the user menu. To change the language, the user performs the following steps:

1. Press the Menu button.

2. Select Language (option 18).

3. Select Edit to change lang uage.

4. Select Option to change the language.

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5. Select OK and Save.

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Dictionaries are currently available for the following language s:

• English

• Spanish

• German,

• Dutch

• Italian

• French

• Portuguese

• Danish

• Swedish

• Czech

• Slovak

For information about configuring the la nguages av ailable t o the user throu gh the LCD m enu, refer to the

“Localization” section on page 3-6.

18 Language (SPA922, 942, and 962)

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SPA900 Series Phone Field Reference

This chapter describes the fields within each section of the following administration web server pages:

• Info Tab, page 5-1

• System Tab, page 5-5

• SIP Tab, page 5-8

• Regional Tab, page 5-17

• Ext 1/2/3/4/5/6 Tab, page 5-36

• Phone Tab, page 5-31

• Ext 1/2/3/4/5/6 Tab, page 5-36

• User Tab, page 5-48

For information about the Provisioning tab, see the Linksys SPA Provisioning Guide.

Info Tab

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• System Information, page 5-1

• Product Information, page 5-2

• Phone Status, page 5-2

Note The fields on this tab are read-only and cannot be edited.

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Info Tab

System Information

Field Description

DHCP Indicates if DHCP is enabled. Current IP Displays the current IP address assigned to the 900 Series pho ne. Host Name Displays the current IP address assigned to the 900 Series phone. Domain Displays the network domain name of the 900 Series phone. Current Netmask Displays the network mask assigned to the 900 Series phone. Current Gateway Displays the default router assigned to the 900 Series phone. Primary DNS Displays the primary DNS server assigned to the 900 Series phone. Secondary DNS Displays the secondary DNS server assigned to the 900 Series phone.

Product Information

Field Description

Product Name Model number of the 900 Series phone. Serial Number Serial number of the 900 Series phone. Software Version Version number of the 900 Series phone software. Hardware Version Version number of the 900 Series phone hardware. MAC Address MAC address of the 900 Series phone. Client Certificate Status of the client certificate, which authenticates the 900 Series phone for use

in the ITSP network. This field indicates if the client certificate is properly installed in the 900 Series phone.

Customization For an RC unit, this field indicates whether the unit has been customized or not.

Pending indicates a new RC unit that is ready for provisioning. If the unit has already retrieved its customized profile, this field displays the name of the company that provisioned the unit.

Licenses Indicates any addition al licenses that you have installed in the 900 Series phone.

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Specifications and Main Features

Frequently Asked Questions

User Manual

Linksys 900 Series IP Phone Administrator Guide

CONTENTS

Preface

Document Audience

Linksys 900 Series IP Telephones

How This Document is Organized

Document Conventions

Related Documentation

Technical Support

Overview

Introducing Linksys 900 Series IP Phones

SPA900 Series Features

SPA901 Features

SPA92x, SPA94x, and SPA962 Features

Ensuring Voice Quality

Feature Descriptions

SIP Proxy Redundancy

Supported Codecs

Other Features

Technology Background

Session Initiation Protocol

Using 900 Series Phones with a Firewall or Router

Network Address Translation

NAT Overview

NAT Types

Simple Traversal of UDP Through NAT

SIP-NAT Interoperation

Where to Go From Here

Getting Started

Linksys 900 Series IP Phones

Caring for Your Hardware

SPA901

Front Panel and Side of Phone

Back Panel

SPA92x, SPA94x, and SPA962 Hardware Features

SPA921

Front Panel

Back Panel

SPA922

SPA941

Front Panel

Back Panel

SPA942

SPA962

Front Panel

Back Panel

Establishing Connectivity

Bandwidth Requirements

Installing the SPA900 Series IP Phone

Assembling the Phone and Connecting to the Network

Attaching the Desk Stand

Mounting the Phone to the Wall

Turning on the Phone

Using the Administration Web Server

Connecting to the Administration Web Server

Administrator Account Privileges

Web Interface URLs

Upgrade URL

Provisioning

Resync URL

Reboot URL

Provisioning Capabilities

Configuration Profile

Using the Interactive Voice Response Interface

Using the IVR Menu

IVR Options

Entering a Password through the IVR

Managing Linksys 900 Series IP Phones

Using the LCD Display

LCD Display Controls

Using Soft Keys

Entering and Saving Settings

Localization

Changing the Display Background (SPA962)

Call Appearances and Extensions

Line Key LEDs