Linksys SPA 900 User Manual 2

Linksys Voice System SPA 900 Series IP Phones

ADMINISTRATION GUIDE

BUSINESS SERIES

Document Audience i-ix

How This Document is Organized i-x

Preface

This guide describes administration and use of the Linksys Voice System (L VS) IP phones. This p reface describes:

• Document Audience, page ix

• How This Document is Organized, page x

• Related Documentation, page x

• Technical Support, pa ge x

Document Audience

This document is written for the following audience:

• Internet Telephony Service Providers (ITSPs, abbreviated to SPs) offering services using LVS

products

• Value-Added Resellers (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 SPs. Please

consult with your SP for specific parameters.

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How This Document is Organized

To do this ... Refer to

Get an overview of Linksys IP phones and their functions

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

Configure and monitor a Linksys 900 Series IP phone

Find configuration and monitoring options available from the LCD display on SPA900 Series IP phones

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

Preface

Chapter 1, “Introducing Linksys IP Phones”

Chapter 2, “Getting Started”

Chapter 3, “Managing Linksys IP Phones”

Chapter 4, “SPA900 Series LCD Command Reference”

Chapter 5, “Linksys IP Phone Field Reference”

Technical Support

If you are an end user of LVS products and need technical support, contact the VAR or SP 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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Overview

CHA P T E R

Introducing Linksys IP Phones

This chapter introduces the functionality of the Linksys 900 Series IP phones and includes the following :

• Overview, page 1-1

• Linksys IP Phone Features, page 1-2

• Feature Descriptions, page 1-5

• Technology Background, page 1-8

Table 1-1 summarizes the ports and features provided by the Linksys 900 Series IP phones described in

this document.

Table 1-1 Linksys IP Phones

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Product Name RJ-45 Voice Lines Additional Features/Notes

SPA901 One (1) One (1) No display. SPA921 One (1) One (1) 1 display. SPA922 Two (2) One (1) Monochrome display, Power over Ethernet (PoE)

support, and an extra Ethernet port for connectin g another device to the LAN.

SPA932 — — Attendant sidecar for SPA962 with 32

LEDs/buttons for monitoring and call t ransfer. Support for Broadsoft Busy Lamp Field and

Asterisk Line Monitoring. SPA941 One (1) Four (4) Monoch rom e displ ay. SPA942 Two (2) Four (4) Monochrome display, Power over Ethernet (PoE)

support, and an extra Ethernet port for connecting

another device to the LAN SPA962 Two (2) Six (6) High-resolution color display, Power over Ethernet

(PoE) support, and an extra Ethernet port for con-

necting another device to the LAN

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Linksys IP Phone Features

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

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

Figure 1-1 Linksys IP Phones in a VoIP Network

Chapter 1 Introducing Linksys IP Phones

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

Linksys IP Phone Features

The following telephony features are provided by the differen t models of Linksys IP phon es. An asterisk (*) indicates that the feature requires support by the SIP server.

• Shared Line Appearance *

–

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

–

WIP310, SPA921, and SPA922: Two call appearances

–

SPA941 and SPA942: Four call appearances

–

SPA962: Six call appearances

• Line Status Indicators

• Call Hold

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• Music on Hold *

• Call Waiting

• Outbound Caller ID Blocking

• Call Transfer - Attended and Blind

• 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

Linksys IP Phone Features

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

• Speed Dialing

• Automatic Redial

• Configurable Dial/Numbering Plan Support - per Line

• 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

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Linksys IP Phone Features

SPA901 Features

The SP A901 prov ides the following features that are not needed with the Linksys IP phon es that prov ide an LCD display:

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

• Ringer and Handset Vo lume Controls

• Handset Input Gain Adjustment

SPA92x, SPA94x, SPA962

The SPA921, SPA922, SPA941, SPA942, and SPA962 provide an LCD display and ad ditional featur es 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

Chapter 1 Introducing Linksys IP Phones

• 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

• 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 T elephony service shoul d be indistingui shable 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 be tter.

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

information about bandwidth requirements for each supported codec, refer to Table 2-1 on page 2-10.

Table 1-2 Speech Quality Metrics

1-4

Algorithm Complexity MOS Score

G.71 1 Very low G.726 Low 4.1 (32 kbps) G.729a Low–medium 4

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Table 1-2 Speech Quality Metrics

G.729 Medium 4 G.723.1 High 3.8

Note Linksys IP phones support all the above voice coding algorithms.

Feature Descriptions

Linksys 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 understand ing of the feature bread th and cap abilities o f Linksys IP p hones.

• SIP Proxy Redundancy, page 1-5

• Supported Codecs, page 1-6

• Other Features, page 1-7

Feature Descriptions

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. Linksys IP phones support the use of backup SIP proxy servers so that service disruption should be nearly eliminated.

A simple way to support proxy redundancy is to configure a SIP proxy server in the Linksys IP phone configuration profile.

a. In Linksys SPA Configuration menu web GUI, enter your service provider name in the Proxy field.

The system

b. In the DNS SRV Auto Prefix filed, enter Yes. c. In the User DNS SRV field, enter Yes. d. The phone tries to register and the server sends a list of IP addresses in order of priority.

(automatically)

where the list is arranged in order of priority. The Linksys IP phone attempts to contact the highest priority proxy server whenever possible.

The dynamic nature of SIP message routing makes the us e 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 Linksys IP phone. One solution to this situation is through the use of DNS SRV records. Linksys IP phones can be instructed to contact a SIP proxy server in a domain named in SIP messages. The Linksys 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 Linksys IP phone tries to contact the list of hosts in the order of their stated priority.

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Feature Descriptions

If the Linksys 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 Linksys IP p hone to “match” a codec name with the far-end device/gateway codec name. Linksys 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.

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 IP phones.

Table 1-3 Codecs Supported by Linksys IP Phones

Codec (Voice Compression Algorithm) Description

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

G.726 This low complexity codec supports compressed 16, 24, 32, and 40

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

G.723.1 Linksys IP phones support the use of ITU G.723.1 audio codec at

Chapter 1 Introducing Linksys IP Phones

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.

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.

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.

1-6

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

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Other Features

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

Table 1-4 Linksys Phone Features

Feature Description

Music On Hold On a connected call, Linksys IP phones may place the rem ote pa rty on ca ll .

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

Adjustable Audio Frames Per Packet

DTMF In-Band and Out-of-Band (RFC 2833) (SIP INFO) Lin ksys IP phon es may

Call Progress Tone Generation

Call Progress T one Pass Through

Jitter Buffer—Dynamic (Adaptive)

Feature Descriptions

If the remote party indicates that they can still recei ve audio while t he call is holding, the MOH server sends streaming audio.

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

This feature allows you to set the number of audio frames contained in one RTP 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.

relay DTMF digits as out-of-band events to preserv e 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.

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

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

Linksys 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, depen ding on changing network conditions.

Linksys IP phones have a Network Jitter Level control setting for each line of service. The jitter level decides how aggressively Linksys 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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Technology Background

Table 1-4 Linksys Phone Features (continued)

Feature Description

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.

Linksys IP phones have three configurable interdigit timers:

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

Chapter 1 Introducing Linksys IP Phones

Report Generation and Event Logging

Syslog and Debug Server Records

Dynamic Payload When no static payload value is assigned per RFC 1890, Linksys IP phones

Call Statistics and Reporting

Technology Background

• 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.

Linksys 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 message-digest authentication, for instance. The information may be organized as an XML page or HTML page.

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

can support dynamic payloads for G.726. The statistics collected by Linksys IP phones during normal operation

statistics are available in the Info tab. Line status is reported fo r each line (1 and 2). Each line maintains up to 2 calls: Call 1 and 2.

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This section provides background information about the technology and protocols used by the phone:

• Session Initiation Protocol, page 1-9

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• Using Linksys IP Phones with a Firewall or Router, page 1-9

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. Figure 1-2 illustrates a 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

SIP UA

Technology Background

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 resp onse 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 Linksys IP Phones with a Firewall or Router

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

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

• RTP ports—16384 to 16482

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Technology Background

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 1-10

• NAT Types, page 1-11

• Simple Traversal of UDP Through NAT (STUN), page 1-11

• SIP-NAT Interoperation, page 1-12

NAT Overview

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, that 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 Se ssion Border Controller (s ee Figure 1-3).

Chapter 1 Introducing Linksys IP Phones

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

External IP address assigned by ISP

ISP

Internet

SPA9000

ITSP

SIP Proxy

192.168.1.100

Session Border

Controller

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

A typical application of NAT allows all the devices in a subscriber 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.

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NAT Types

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 internal 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 external 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. I f 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 (STUN)

Technology Background

Simple Traversal of User Datagram Protocol (UDP) through Network Address Translators (NATs), or STUN, is a protocol defined by RFC 3489 that allows a client behind a NAT device to determine its public address, the type of NAT implemented, 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:

Enable debugging on the Linksys IP 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).

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 Linksys 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.

5. To determine the type of NAT your router is using set <STUN Test Enable> to yes.

6. View the debug messages to determine if your network uses symmetric NAT.

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Technology Background

SIP-NAT Interoperation

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

Note The ITSP might offer an outbound NAT-Aware proxy, which discovers the public IP address from the

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

The Linksys IP phone system must substitute the private IP address information with the pro per external IP address/port in the mapping chosen by the underlying NAT to communicate with a particular public peer address/port. For this, the Linksys IP 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 Linksys IP 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.

Chapter 1 Introducing Linksys IP Phones

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 period , the Linksys IP phone sy stem continues to send periodic keep-alive packets through the mapping to extend its validity as necessary.

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CHA P T E R

Getting Started

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

• Linksys 900 Series IP Phones, page 2-1

• SPA9000 IP PBX System, page 2-9

• Establishing Connectivity, page 2-9

• Using the Administration Web Server, page 2-12

• Web Interface URLs, page 2-14

• Provisioning, page 2-15

• Using the Interactive Vo ice Response Interfac e, page 2-1 6

Note If the Linksys IP phone is supplied or sponsored b y 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 provides fully-featured VoIP phones that integrate with the Linksys SPA9000 to provide connectivity to other local stations, and through an ITSP to IP phones over the Internet, In addition, t he 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 Hardwa re, page 2-2

• SPA901, page 2-2

• SPA92x, SPA94x, and SPA962 Hardware Features, page 2-3

• SPA922, page 2-5

• SPA932, page 2-5

• SPA941, page 2-6

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Linksys 900 Series IP Phones

• SPA942, page 2-7

• SPA962, page 2-8

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 the 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 that can be w all 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

Chapter 2 Getting Started

• Ethernet LAN – 10BaseT RJ-45

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

Figure 2-1 SPA901

The following tables describe the status indicators and controls on the front of the device and the port s 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: shared

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

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

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

three-way conference call.

SPA92x, SPA94x, and SPA962 Hardware Features

The 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 (six LED’s on the SPA962)

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

• Line LED Configurable to 13 Different States (See “Line Key LED Pattern” section on page 5-31).

• 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 Retrieval Button

• 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

• Standard 12-Button Dialing Pad

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• High Quality Handset and Cradle

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SPA921

Chapter 2 Getting Started

• Built-In High Quality Microphone and Speaker

• Headset Jack – 2.5 millimeter

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

• 802.3af Compliant Power over Ethernet (PoE) (not available on SPA9x1 models)

• 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).

Figure 2-2 SPA921

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 SPA9000 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 Figure 2-3). The SPA100 power supply must be ordered separately if you are not using a PoE switch.

Figure 2-3 SPA922

SPA932

Firmware Version 5.1

The SPA932 is a 32-button attendant console for the SPA962, providing 32 three-color (red, green, and amber) programmable LEDS, with support for Broadsoft Busy Lamp Field and Asterisk Line Monitoring (see Figure 2-4). The SPA932 attaches to the SPA962 with the attachment arm provided (not shown). It obtains power directly from the SPA962 and does not require a separate power supply. Two SPA932 units can be attached to a single SPA962 to monitor a total of 64 separate lines.

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Linksys 900 Series IP Phones

Figure 2-4 SPA932 (Front View with SPA962)

Each SPA932 unit provides 32 programmable speed-dial or direct station select (DSS) buttons, Each lighted button indicates line status (idle, ringing, busy, or null) using a busy lamp field (BLF) . Incoming calls can be immediately transferred to the proper location by pushing a button assigned to the exten sion on the SPA932.

The following table summarizes the meaning of each light color and pattern.

Chapter 2 Getting Started

SPA941

Port/LED Meaning

AUX IN Connects to the SPA962 AUX OUT Connects to a second SPA932 unit (optional) Green Idle Red In-use Blinking Red Ringing

The SPA941 provides four lines (see Figure 2-5).

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Firmware Version 5.1

Chapter 2 Getting Started

Linksys 900 Series IP Phones

Figure 2-5 SPA941

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

Front Panel

Back Panel

SPA942

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 SPA9000 through a local switch. Power Connects to the 5-volt power supply.

Firmware Version 5.1

The SPA942 is similar to the SPA941, but provides two Ethernet ports for connecting to the LAN and supports Power over Ethernet (see Figure 2-6). The PA100 power supply must be ordere d separately if you are not using a PoE switch. See the table for SPA941.

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+ 135 hidden pages

Linksys SPA 900 User Manual 2

Specifications and Main Features

Frequently Asked Questions

User Manual

Preface

Document Audience

Related Documentation

Technical Support

Overview

Introducing Linksys IP Phones

Linksys IP Phone Features

SPA901 Features

SPA92x, SPA94x, SPA962

Ensuring Voice Quality

Feature Descriptions

SIP Proxy Redundancy

Supported Codecs

Other Features

Technology Background

Session Initiation Protocol

Using Linksys IP Phones with a Firewall or Router

Network Address Translation

NAT Overview

NAT Types

Simple Traversal of UDP Through NAT (STUN)

SIP-NAT Interoperation

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

SPA932

SPA941

Front Panel

Back Panel

SPA942