Linksys IP Phone SPA 900 User Manual

ADMINISTRATION GUIDE  
BUSINESS SERIES  
Linksys Voice System  
SPA 900 Series IP Phones  
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Contents  
User 5-48  
General 5-51  
Unit 1 5-52  
Unit 2 5-53  
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Preface  
This guide describes administration and use of the Linksys Voice System (LVS) IP phones. This preface  
describes:  
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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Preface  
How This Document is Organized  
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 Chapter 2, “Getting Started”  
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 Reference”  
IP phones  
Find the function and usage for each field or  
parameter on the Linksys 900 Series IP phone  
administration web server pages  
Related Documentation  
The following documentation provides additional information about Linksys IP phones:  
SPA IP Phone User Guide  
SPA IP Phone Quick Reference Guide  
SPA Provisioning Guide (Linksys Partner Login required)  
Linksys Voice System Installation Guide  
Linksys Voice over IP Product Guide: SIP CPE for Massive Scale Deployment  
SPA 2.0 Analog Telephone Adapter Administrator Guide  
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  
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C H A P T E R  
1
Introducing Linksys IP Phones  
This chapter introduces the functionality of the Linksys 900 Series IP phones and includes the following:  
Overview  
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  
Product Name RJ-45  
Voice Lines  
One (1)  
Additional Features/Notes  
No display.  
SPA901  
SPA921  
SPA922  
One (1)  
One (1)  
Two (2)  
One (1)  
1 display.  
One (1)  
Monochrome display, Power over Ethernet (PoE)  
support, and an extra Ethernet port for connecting  
another device to the LAN.  
SPA932  
Attendant sidecar for SPA962 with 32  
LEDs/buttons for monitoring and call transfer.  
Support for Broadsoft Busy Lamp Field and  
Asterisk Line Monitoring.  
SPA941  
SPA942  
One (1)  
Two (2)  
Four (4)  
Four (4)  
Monochrome display.  
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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Chapter 1 Introducing Linksys IP Phones  
Linksys IP Phone Features  
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.  
Figure 1-1 illustrates how the IP phones are connected in a VoIP network.  
Figure 1-1 Linksys IP Phones in a VoIP Network  
Linksys IP Phone Features  
The following telephony features are provided by the different models of Linksys IP phones. 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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Linksys IP Phone Features  
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  
Ten-User Downloadable 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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Chapter 1 Introducing Linksys IP Phones  
Linksys IP Phone Features  
SPA901 Features  
The SPA901 provides the following features that are not needed with the Linksys IP phones that provide  
an LCD display:  
Built-in Interactive Voice Response (IVR) system to check status and change configuration  
Ringer and Handset Volume Controls  
Handset Input Gain Adjustment  
SPA92x, SPA94x, SPA962  
The SPA921, SPA922, SPA941, SPA942, and SPA962 provide an LCD display and additional 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 Blocking  
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  
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. For  
information about bandwidth requirements for each supported codec, refer to Table 2-1 on page 2-10.  
Table 1-2 Speech Quality Metrics  
Algorithm  
G.711  
Complexity  
Very low  
Low  
MOS Score  
4.5  
G.726  
4.1 (32 kbps)  
4
G.729a  
Low–medium  
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Feature Descriptions  
Table 1-2 Speech Quality Metrics  
G.729  
Medium  
High  
4
G.723.1  
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 understanding of the feature breadth and capabilities of Linksys IP phones.  
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 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 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 phone 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  
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.  
G.726  
This low complexity codec supports compressed 16, 24, 32, and 40  
kbps digitized voice transmission at one through ten 10 ms voice  
frames per packet. This codec provides high voice quality.  
G.729A  
The ITU G.729 voice coding algorithm is used to compress  
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  
Linksys 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, Linksys IP phones can support dynamic  
payloads for G.726.  
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Feature Descriptions  
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 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 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.  
DTMF  
In-Band and Out-of-Band (RFC 2833) (SIP INFO) Linksys 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  
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.  
Call Progress Tone Pass This feature allows you to hear the call progress tones (such as ringing) that  
Through are generated from the far-end network.  
Jitter Buffer—Dynamic Linksys IP phones can buffer incoming voice packets to minimize  
(Adaptive)  
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.  
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  
Voice Activity Detection (VAD) with Silence Suppression is a means of  
Detection with Silence increasing the number of calls supported by the network by reducing the  
Suppression and  
Comfort Noise  
Generation  
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.  
Configurable Dial Plan Linksys IP phones have three configurable interdigit timers:  
with 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.  
Report Generation and Linksys IP phones report a variety of status and error reports to assist service  
Event Logging  
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.  
Syslog and Debug  
Server Records  
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.  
Dynamic Payload  
When no static payload value is assigned per RFC 1890, Linksys IP phones  
can support dynamic payloads for G.726.  
Call Statistics and  
Reporting  
The statistics collected by Linksys 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.  
Technology Background  
This section provides background information about the technology and protocols used by the phone:  
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Technology Background  
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  
2
4
SIP Proxy  
RTP  
SIP Proxy  
3
SIP Proxy  
1
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 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  
If security is not a concern in your environment, you can consider disabling SPI, if this function exists  
on your firewall.  
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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  
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 Session Border Controller (see Figure 1-3).  
Figure 1-3 NAT Support with Session Border Controller Provided by ITSP  
Private IP address  
192.168.1.1  
External IP address  
assigned by ISP  
192.168.1.101  
192.168.1.102  
NAT Device  
Internet  
ISP  
DHCP  
server  
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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Technology Background  
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. 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 (STUN)  
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:  
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 not have firewall running on your PC that could block the syslog port (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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Chapter 1 Introducing Linksys IP Phones  
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 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 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.  
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 system continues to send  
periodic keep-alive packets through the mapping to extend its validity as necessary.  
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C H A P T E R  
2
Getting Started  
This chapter describes the tools and utilities available for administering Linksys phones. It includes the  
following sections:  
Note  
If the Linksys 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 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, 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:  
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Chapter 2 Getting Started  
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 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 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  
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: shared  
line on hold. On red: line in use. Flashing red: local call on hold. On  
orange: phone not registered.  
FLASH  
Press to pickup a second incoming call, transfer a call, or setup a  
three-way conference call.  
REDIAL  
Press the redial the last number called.  
Back Panel  
Port  
Function  
Phone jack  
Ethernet port  
Power  
Connect to the handset.  
Connect to the SPA9000 through a local switch.  
Connect to the 5-volt power supply.  
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  
High Quality Handset and Cradle  
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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  
SPA921  
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  
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.  
Front 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  
Soft keys 1-4  
Scrolls between display and configuration options in the LCD  
display.  
Selects options on the LCD display  
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Back Panel  
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.  
SPA922  
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  
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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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 extension  
on the SPA932.  
The following table summarizes the meaning of each light color and pattern.  
Port/LED  
AUX IN  
AUX OUT  
Green  
Meaning  
Connects to the SPA962  
Connects to a second SPA932 unit (optional)  
Idle  
Red  
In-use  
Ringing  
Blinking Red  
SPA941  
The SPA941 provides four lines (see Figure 2-5).  
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Figure 2-5 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.  
Front Panel  
Feature  
Function  
LCD display  
Telephone keypad  
Lists device status and configuration options.  
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.  
Displays status of each extension.  
Line status indicators 1-4  
Back Panel  
Port  
Function  
Phone jack  
Ethernet ports  
Power  
Connects to the handset.  
Connects to the SPA9000 through a local switch.  
Connects to the 5-volt power supply.  
SPA942  
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 ordered separately if  
you are not using a PoE switch. See the table for SPA941.  
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Figure 2-6 SPA942  
SPA962  
The SPA962 has a 320 x 240 true color, four-inch, LCD, provides up to six telephone extensions, and  
supports PoE (see Figure 2-7). The PA100 power supply must be ordered separately if you are not using  
a PoE switch.  
Figure 2-7 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.  
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SPA9000 IP PBX System  
Front 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  
Soft keys 1-4  
Scrolls between display and configuration options in the LCD display.  
Selects options on the LCD display.  
Line status indicators 1-6  
Displays status of each extension.  
Back Panel  
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.  
Connects to the 5-volt power supply.  
Connects to optional SPA932  
Power  
AUX RJ-11 port  
SPA9000 IP PBX System  
The SPA9000 IP PBX System (VoIP side), along with the SPA400, which is Internet Telephony  
Gateway, provides for an analog line and voicemail.  
The system provides auto attendant features for multiple extensions. Features include:  
Auto Attendant  
Shared line appearances  
Configurable call routing  
Multiple DID numbers per VoIP line  
Call hunting (sequential, round robin, random)  
Call transfer and Call forward  
Call hold and Call waiting  
You can perform SPA9000 administrative tasks using an Interactive Voice Response (IVR) system or a  
built-in Web Server.  
Establishing Connectivity  
This section describes how to connect the SPA900 Series IP phone hardware. It includes the following  
topics:  
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Establishing Connectivity  
Bandwidth Requirements1  
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 Linksys IP phone with QoS-capable networking equipment that can  
prioritize the VoIP application traffic. QoS features are available on many Linksys data networking  
switches and routers. A QoS-enabled router prioritizes the packets going upstream to the ISP. Table 2-1  
illustrates the bandwidth budget using different codecs.  
Table 2-1 Ethernet Bandwidth Budget for Off-Net VoIP Calling  
Approximate bandwidth budget for  
each side of conversation  
Codec  
2 calls  
4 calls  
6 calls  
8 calls  
G.711  
110 kbps  
220 kbps 440 kbps 660 kbps 880 kbps  
174 kbps 348 kbps 522 kbps 696 kbps  
158 kbps 316 kbps 474 kbps 632 kbps  
142 kbps 284 kbps 426 kbps 568 kbps  
126 kbps 252 kbps 378 kbps 504 kbps  
110 kbps 220 kbps 330 kbps 440 kbps  
G.726-40 87 kbps  
G.726-32 79 kbps  
G.726-24 71 kbps  
G.726-16 63 kbps  
G.729  
55 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 websites:  
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)  
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Establishing Connectivity  
Assembling the Phone and Connecting to the Network  
Perform the following steps to connect Linksys IP 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  
Step 3  
Plug in the short straightened end of the handset cord into the jack at the bottom of the handset.  
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  
Step 5  
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.  
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 on the SPA962 is reserved for SPA 932 use.  
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  
Step 2  
Step 3  
Line up the tabs on the desk stand with the slots on the back of the phone.  
Slide the bottom tabs into the slots  
Lightly press down on the top of the desk stand. It should easily slide into the top slots. Do not force.  
Mounting the Phone to the Wall  
Note  
When mounting the phone to a wall, do not attach the desk stand.  
Step 1  
Step 2  
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.  
Push down slightly to lock the phone in place.  
You can purchase an optional wall mount kit (MB100) from your retailer or Linksys.com.  
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Using the Administration Web Server  
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.  
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 seconds, the phone displays 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  
Using the Administration Web Server  
This section describes how to use the administration web server to configure the Linksys IP phone. It  
includes the following topics:  
Connecting to the Administration Web Server  
To access the Linksys IP phone administration web server, perform the following steps.  
Step 1  
Step 2  
Launch a web browser on a computer connected to the Linksys IP phone .  
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 follow these steps:  
a. Press **** on the keypad to access the IVR menu.  
b. Press 110# to determine the Internet (WAN) IP address.  
Direct the browser to the IP address of the Linksys IP phone.  
Step 3  
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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 them.  
Note  
Changing between the Router and Voice pages discards any unsubmitted changes to either page.  
Step 4  
Step 5  
Click the Voice tab.  
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  
Step 7  
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.  
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 Linksys IP phone  
may reboot.  
Administrator Account Privileges  
The Linksys IP phone support two levels of administration privileges: Administrator and User. Both  
privileges can be password protected.  
Note  
By default, there are no passwords assigned for either the Administrator account or the User account.  
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:  
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.  
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Web Interface URLs  
Web Interface URLs  
The Linksys IP phone web interface supports several functions through special URLs:  
Upgrade  
Reboot  
Resync  
Administrator account privilege is needed for these functions.  
Upgrade URL  
The Upgrade URL lets you upgrade the Linksys IP 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 upgrade the  
Linksys IP phone even if the web page indicates otherwise.  
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 upgrade operation.  
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  
Resync URL  
The Resync URL lets you force the Linksys IP phone to do a resync to a profile specified in the URL,  
which can identify either a TFTP, HTTP, or HTTPS server.  
Note  
The SPA resyncs only when it is idle.  
The syntax of the Resync URL is as follows:  
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  
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Provisioning  
Reboot URL  
The Reboot URL lets you reboot the Linksys IP phone.  
Note  
The Linksys IP phone reboots only when it is idle.  
Provisioning  
This section describes the provisioning functionality of the Linksys IP phone. This section includes the  
following topics:  
For detailed information about provisioning the Linksys IP phone, refer to the Linksys SPA Provisioning  
Guide.  
Provisioning Capabilities  
The Linksys IP phone provides for secure provisioning and remote upgrade. Provisioning is achieved  
through configuration profiles transferred to the device via TFTP, HTTP, or HTTPS.  
The Linksys IP 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 Linksys IP phone accepts profiles in XML format, or alternatively in a proprietary binary format,  
which is generated by a profile compiler tool, SIP Profile Compiler (SPC), available from Linksys. The  
Linksys IP phone supports up to 256-bit symmetric key encryption of profiles. For the initial transfer of  
the profile encryption key (initial provisioning stage), the Linksys IP 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 Linksys IP 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  
Linksys IP phone via a remote profile resync.  
For further information about remote provisioning refer to the LVS SPA Provisioning Guide.  
Configuration Profile  
The Linksys IP 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:  
<flat-profile>  
<Admin_Passwd>some secret</Admin_Passwd>  
<Upgrade_Enable>Yes</Upgrade_Enable>  
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Using the Interactive Voice Response Interface  
</flat-profile>  
Binary format profiles contain Linksys IP 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 tool:  
Admin_Passwd “some secret”;  
Upgrade_Enable “Yes”;  
Refer to the LVS SPA Provisioning Guide for further details.  
The names of parameters in XML profiles can generally be inferred from the Linksys IP 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], or [4]) are 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 900  
Series LCD Display” section on page 3-1) or the administration web server (see the “Using the  
This section includes the following topics:  
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:  
Select the Products > Business > IPCommunications > Linksys Voice System > SPA9000 > IVR Quick  
Reference Card link.  
Using the IVR Menu on a Linksys SPA901 Phone  
To use the IVR menu:  
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Using the Interactive Voice Response Interface  
Step 1  
Step 2  
On a SPA901rt of the Linksys IP phone.  
Press **** (quickly press the star key four times).  
Wait until you hear “Linksys configuration menu.”  
Note  
You cannot access the IVR from a phone that is currently connected to a call.  
Step 3  
Step 4  
Refer to Table 2-2 to identify the option required.  
Enter the required option followed by the # (pound) key.  
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. To save the new setting, press 1. To 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 Linksys IP phone times out. You need to re-enter  
the menu by pressing ****.  
Step 5  
To exit the menu, hang up the telephone.  
The settings you have saved take effect after you hang up the telephone. The Linksys IP 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  
Check DHCP  
3948  
100  
None  
None  
IVR announces whether DHCP is enabled  
or disabled.  
Enable/Disable DHCP  
101  
Enter 1 to enable  
Enter 0 to disable  
Requires password  
Check WAN IP Address 110  
None  
IVR announces the current IP address of  
the WAN port.  
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Using the Interactive Voice Response Interface  
Table 2-2 IVR Options (continued)  
Set Static IP Address  
111  
Enter IP address using  
numbers on the telephone  
DHCP must be “Disabled,” otherwise you  
hear, “Invalid Option,” if you try to set  
key pad. Use the * (star) key this value.  
when entering a decimal  
Requires password  
point.  
Check Network Mask  
Set Network Mask  
120  
121  
None  
IVR announces the current network mask  
of SPA.  
Enter value using numbers on DHCP must be “Disabled,” otherwise you  
the telephone key pad. Use hear, “Invalid Option,” if you try to set  
the * (star) key when entering this value.  
a decimal point.  
Requires password  
Check Static Gateway IP 130  
Address  
None  
IVR announces the current gateway IP  
address of SPA.  
Set Static Gateway IP  
Address  
131  
Enter IP address using  
numbers on the telephone  
DHCP must be “Disabled,” otherwise you  
hear, “Invalid Option,” if you try to set  
key pad. Use the * (star) key this value.  
when entering a decimal  
Requires password  
point.  
Check MAC Address  
140  
None  
IVR announces the MAC address of SPA  
in hex string format.  
Check Firmware Version 150  
None  
None  
IVR announces the version of the  
firmware running on the SPA.  
Check Primary DNS  
Server Setting  
160  
IVR announces the current setting in the  
<Primary DNS> parameter.  
Set Primary DNS Server 161  
Enter IP address using  
numbers on the telephone  
key pad. Use the * (star) key  
when entering a decimal  
point.  
Requires password  
Check administration web 170  
server port  
None  
IVR announces the port that the web  
server is listening on. (Default is 80)  
Check LAN IP Address  
210  
180  
181  
None  
IVR announces the current IP address of  
the LAN port.  
Check PBX multicast  
address  
None  
IVR announces the current value.  
Set PBX multicast  
address  
Enter IP address and port.  
Use * key for entering a dot. Port fields. Requires password  
For example,  
Enter a * between the IP address and the  
224.168.168.169:8089 is  
224*168*168*169*8089.  
Enable/Disable  
administration web server  
7932  
Enter 1 to enable  
Enter 0 to disable  
Requires password  
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Using the Interactive Voice Response Interface  
Table 2-2 IVR Options (continued)  
Manual Reboot of Unit  
732668  
877778  
None  
After you hear “Option Successful,” hang  
up. Unit reboots automatically.  
User Factory Reset of  
Unit  
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.  
WARNING:  
ALL “User-Changeable”  
NON-DEFAULT  
SETTINGS WILL BE  
LOST!  
This might include  
network and service  
provider data.  
Factory Reset of Unit  
WARNING:  
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 configuration parameters are reset to  
factory default values.  
ALL NON-DEFAULT  
SETTINGS WILL BE  
LOST!  
This includes network and  
service provider data.  
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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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 configuration 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 necessary, the Linksys IP phone  
automatically reboots.  
2. After one minute of inactivity, the unit times out. The user needs to re-enter the configuration menu  
from the beginning by pressing * * * *.  
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C H A P T E R  
3
Managing Linksys IP Phones  
This chapter describes how to configure Linksys IP phones and includes the following sections:  
Using the 900 Series LCD Display  
All Linksys IP phone models provide an LCD display except for the SPA901, which you configure using  
the Interactive Voice Response (IVR) interface, described in the “Using the Administration Web Server”  
section on page 2-12. This section describes the basic functions of the LCD display and includes the  
following topics:  
SPA900 Series LCD Display Controls  
The LCD display on the SPA900 Series IP phones provides menu options for operating, configuring and  
monitoring the IP phone. For information about each of these options, refer to Chapter 4, “SPA900  
Series LCD Command Reference”. Figure 3-1 illustrates the main controls on the LCD display.  
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Using the 900 Series LCD Display  
Figure 3-1 SPA900 Series LCD Controls  
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.  
Figure 3-2 summarizes the function of the other buttons on the telephone keypad.  
Figure 3-2 Telephone Keypad  
Voicemail  
Call hold  
Setup  
Microphone mute  
Headset  
Volume control  
Speaker  
Numeric keypad  
To access an option, press the Setup button and press the Navigation button to select the option. Then  
push the left soft key beneath the Select option to select the option. To use the shortcut to an option,  
press the 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.  
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Using the 900 Series LCD Display  
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.  
Typically, 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.  
Note  
The following table shows the soft key labels for the SPA962 in parentheses ( ). This manual generally  
shows procedures using the non-SPA962 labels.  
Table 3-1 General LCD Display Soft Key Labels  
Label (SPA962 Label)  
Description  
cfwd (Forward)  
Forwards calls, with one of four options: All, When the line is busy,  
When there is no answer, or When there is no answer after a time  
delay.  
dir (Directory)  
dnd (DND)  
Accesses the phone directory.  
(Do not disturb) Sends incoming calls to voicemail, or gives a busy  
signal if voicemail is not available.  
grPick (GrPickup)  
lcr (Call Rtn)  
Picks up a call from an extension outside the group.  
(Last call return) Dials the last call received.  
Picks up a call from another phone in the group.  
Displays a list of recently called numbers.  
Picks up a call from where it was parked.  
pickup (Pickup)  
redial (Redial)  
unpark (Unpark)  
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 Setup  
button. This manual generally shows procedures using the non-SPA962 labels.  
Note  
The following table shows the soft key labels for the SPA962 in parentheses ( ). This manual generally  
shows procedures using the non-SPA962 labels.  
Table 3-2 Setup Soft Keys  
Label (SPA962 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 (Add)  
Adds an entry.  
alpha, IP, num (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 (Cancel)  
Cancels change before pressing save or ok.  
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Using the 900 Series LCD Display  
Table 3-2 Setup Soft Keys  
Label (SPA962 Label)  
change (Change)  
clear (Clear)  
Description  
Changes the ring tone feature.  
Clears the entire entry.  
copy (Copy)  
In a directory, use the Navigation button to highlight an entry, then  
press to copy an entry.  
delChr (DeleteChr)  
delete (Delete)  
dial (Dial)  
Deletes the last character entered.  
Deletes an entry.  
Places a call to the number highlighted on the LCD screen.  
Edits part of an entry.  
edit (Edit)  
ok (OK)  
Confirms an operation.  
paste (Paste)  
Use the Navigation button to highlight an entry in the personal  
directory and press paste to place a copied entry  
play (Play)  
save (Save)  
select (Select)  
y/n (Y/N)  
Plays ring tones.  
Saves changes.  
Chooses an item to change or view.  
Toggles between yes and no settings.  
Entering and Saving Settings  
The SPA900 Series phone GUI lets you enter alphanumeric and numeric inputs. You can enter the  
following data types 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 option or selecting a preferred audio device  
or ring tone  
When more than one data type applies to an entry, the phone shows a soft key for switching input modes  
among the types: you can scroll through the available types, such as num, alpha, or IP. The cursor also  
changes according to Table 3-3, to indicate your current input mode:  
Table 3-3 Input Modes  
Input Mode  
Numeric  
Soft Key  
num  
Cursor (appending)  
Cursor (inserting)  
Blinking underscore _  
Blinking vertical line |  
Alphanumeric  
alpha  
Blinking full-height  
block ¶  
Blinking open rectangle  
.
IP Address  
IP  
Blinking half-height  
block .  
Blinking half rectangle .  
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Localization  
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 you press each digit key. Scroll through the choices by pressing the same key multiple times.  
To accept a symbol as input, do 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.  
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 character  
erase<—Erase the character to the left of the cursor  
clear—Clear the entire entry  
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 SPA900 Series phone automatically  
scrolls the input to the left so that you can continue to enter input. On the other hand, the SPA900 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 SPA900 Series phone displays (y/n); pressing the soft key  
toggles the current input between yes and no.  
When the SPA900 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 you press save or ok before exiting the edit screen for that setting.  
However, there are a few exceptions where you 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.  
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Localization  
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>;  
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 set-  
tings tab, as shown in Figure 3-3.  
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:  
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Changing the Display Background (SPA942/962)  
Note  
You must have an ID, Username, and Password for this site.  
The dictionaries are found in the LVS area of the FW and Tools section. Currently dictionaries are  
available for the following languages:  
English  
Spanish  
German,  
Dutch  
Italian  
French  
Portuguese  
Danish  
Swedish  
Czech  
Slovak  
The end user can change the language of the phone from the user menu, by following these steps:  
1. Press the Setup button.  
2. Select Language (option 9).  
3. Select Edit to change language.  
4. Select Option to change the language.  
5. Select OK and Save.  
Changing the Display Background (SPA942/962)  
This section applies to the SPA 942 and SPA962, which provide the option to create a customized  
background. For the SPA962, you can use a 320 x 240 pixel bitmap (.bmp) image (1, 4, 8, 24, or 32-bit  
color) for the background image that appears on the LCD display. For the SPA942, the image should be  
128 x 48 black and white (1-bit color).  
You can include the definition of the required parameters in the configuration profile for the SPA 942 or  
SPA962, or use the administration web server.  
To use the administration web server to configure a customized image perform the following steps:  
Step 1  
Step 2  
Step 3  
Copy the bitmap (.bmp) image you want to use to a TFTP server that is accessible from the SPA962.  
Connect a browser to the administration web server for the SPA 942 or SPA962.  
Click the Phone tab and refer to the General tab at the top of the page (see Figure 3-4).  
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Using the SPA932 (Sidecar) with the SPA962  
Figure 3-4 Background Picture Parameters  
Step 4  
Step 5  
Choose one of the following options from the Select Logo pull-down selection list to control what  
displays when the phone is starting:  
Default—Uses the default Linksys logo.  
BMP Picture—Uses the .bmp file from the URL in the BMP Picture Download URL field.  
None—No logo used.  
Choose one of the following options from the Select Background Picture pull-down selection list to  
control the background used:  
Default—Uses the default Linksys background.  
BMP Picture—Uses the .bmp file from the URL in the BMP Picture Download URL field.  
Text Logo—Uses the text background in the Text Logo field (SPA942 only).  
None—No background used.  
If you select Text Logo, enter up to 63 characters in in the Text Logo field.  
If you select BMP Picture, enter the URL to the server where the BMP picture is accessible. For  
example:  
tftp://mytftpserver.mydomain.com/pictures/background1.bmp  
The URL can use HTTP, HTTPS, or TFTP, with a maximum length of 255 characters.  
Step 6  
Click Submit All Changes.  
Using the SPA932 (Sidecar) with the SPA962  
The SPA932 is a 32-button attendant console for the 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 extension on the SPA932. Two  
SPA932 units can be attached to a single SPA962 to monitor a total of 64 separate lines.  
To configure the server to use the SPA932, configure each extension that will be monitored by the  
SPA932. The SPA932 is a SIP subscriber in relation to each SIP proxy server, which allows the SPA932  
to receive NOTIFY messages from the SIP proxy that indicate the status of each monitored phone. In  
general, the SIP proxy is identified by its IP address, or possibly through a hostname if DNS is properly  
configured. The way in which each phone and extension is identified is vendor-specific.  
To complete the configuration required to use the SPA932, perform the following steps:  
Step 1  
Complete the physical installation of the SPA932 unit, which connects it to the SPA962.  
For instructions regarding the physical installation of the SPA932 and an introduction to its use, refer to  
the 32 Button Attendant Console for the SPA962 IP Phone Quick Installation Guide.  
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Using the SPA932 (Sidecar) with the SPA962  
Step 2  
Step 3  
Configure the SIP proxy server:  
Configure the SPA932 using the administration web server connected to the SPA962 to which the  
SPA932 is physically attached.  
Configuring the SPA9000 for the SPA932  
The default multicast autoconfiguration assigns each phone with an extension starting with 101.  
Even if you use the default autoconfiguration performed by the SPA9000, you must still enable  
Computer Telephony Integration (CTI) on the SPA9000 administration web server, as shown in the  
following procedure.  
Step 1  
Connect to the administration web server for the SPA9000.  
To use the SPA962 with a SPA932, it must be running Release 5.1.12 or later.  
Step 2  
Step 3  
Step 4  
Step 5  
Click Admin Login on the administration web server page.  
If necessary, enter admin and your password.  
Click Advanced.  
Click the SIP tab.  
In the PBX Phone Parameters section, select yes from the CTI Enable pull-down selection list.  
Figure 3-5 CTI Enable Parameter  
Step 6  
Click Submit All Changes at the bottom of the page.  
Configuring the Broadsoft Server for the SPA932  
The Broadsoft server requires you to assign IP phone users to the Busy Lamp Field (BLF) User Monitor  
List. The Broadsoft server sends updates on the status of each phone on this list to the SPA932, which  
subscribes for this service.  
To configure the Broadsoft server to support the SPA932, complete the following steps:  
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Using the SPA932 (Sidecar) with the SPA962  
Step 1  
Step 2  
Step 3  
On the Broadsoft server, in the user profile used by the SPA962 to register with the Broadsoft server,  
select Assign Services.  
On the Assign Services page, from the Available Services column, move Busy Lamp Field and Phone  
Status Monitoring to the User Services column.  
Define the List URI used by the SPA932 to register for BLF monitoring service with the Broadsoft  
server.  
This value must be entered following the sub= keyword, when configuring each key on the 932 (see  
Note  
If you configure more than one monitored list on the Broadsoft server, use the vid= keyword to  
specify the extension to use for each list.  
Step 4  
Step 5  
On the Phone Status Monitoring page, move users that need to be monitored from the Available User  
columnn to the Monitored User column.  
Click the Add (or Add All) button to move each user to the Monitored Users column.  
The Directory Number (DN) associated with each user account when it is created on the Broadsoft  
Server is shown in parenthesis in the Monitored Users list. You use this DN to identify the specific phone  
assigned to each key on the SPA932.  
Save and enable your configuration changes on the Broadsoft server.  
Configuring the Asterisk Server for the SPA932  
To configure the Asterisk server to allow the SPA932 to register for BLF monitoring, perform the  
following steps:  
Step 1  
Step 2  
Step 3  
Add a context in the extensions.conf file.  
Add a Subscribecontext command to point to the context in the sip.conf file.  
Configure the SPA932 to register with the Asterisk server (see  
The following example context uses "home" for extension 3500. This is entered in the file  
extensions.conf.  
Example 3-1 Asterisk Context for SPA932  
[home]  
exten => 3500,1,Dial(SIP/3500/20)  
exten => 3500,hint,SIP/3500  
exten => 3500,2,Voicemail,u3500  
exten =>3500,1,3,hangup  
...  
In the following example, exteion 3500 is used to add "Subscribecontext" to point to the context. This  
is entered in the file sip.conf.  
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Using the SPA932 (Sidecar) with the SPA962  
Example 3-2 Asterisk Subscribecontext for SPA932  
[3500]  
type=friend  
secret=3500  
callerid="spa3500"<3500>  
nat=no  
context=home  
mailbox=3500  
Subscribecontext=home  
...  
...  
Configuring the SPA932  
Complete the configuration required for the extensions on your SPA9000 IP PBX and obtain the  
following information, which is required to configure the SPA932:  
SPA9000 IP address (or hostname if supported)  
Phone extension numbers  
By default, all LEDs on the SPA932 are assigned to the first configured extension on the SPA962 to  
which it is connected. You need to assign the LEDs to any of the other five extensions on the SPA962  
that you want to monitor using the SPA932.  
To configure the SPA932, perform the following steps:  
Step 1  
Connect to the administration web server for the SPA962 to which the SPA932 is connected.  
To use the SPA962 with a SPA932, it must be running Release 5.1.12 or later.  
Step 2  
Step 3  
Step 4  
Step 5  
Click Admin/Advanced on the administration web server page.  
Click the SIP tab.  
Select yes from the CTI Enable pull-down selection list.  
(optional) Change the SPA962 station name.  
Click the Phone tab and edit the Station Name parameter in the General section.  
Step 6  
Click the SPA932 tab.  
If you do not see the SPA932 tab, check your firmware version, and upgrade as necessary.  
The system displays the page shown in Figure 3-6.  
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Using the SPA932 (Sidecar) with the SPA962  
Figure 3-6 SPA932 Configuration  
Step 7  
Step 8  
Select yes from the Unit 1 Enable pull-down selection list.  
If you have installed two SPA932 units, also select yes from the Unit 2 Enable pull-down selection list.  
Make sure that no is selected on the Test Mode Enable selection list.  
This option, which is useful for testing the SPA932, is disabled by default. You cannot complete the  
software configuration for the SPA932 if this option is enabled.  
SPA9000  
Broadsoft  
Asterisk  
Step 9  
For each extension or user that you want to monitor using the SPA932, enter the configuration script in  
the appropriate field for each key.  
Note  
You can identify the user or extension number in the space provided next to each key before  
proceeding. This will help ensure that you enter the correct information in each field.  
The configuration script is composed of the following keywords, followed by an equal sign (=) and  
separated by semicolons (;):  
fnc—defines which of the following functions are enabled for the specified key (separate more than  
one function with a plus sign (+):  
blf—busy lamp field function used for monitoring line activity  
sd—speed dial function  
cp—call pickup (if supported by the SIP proxy server)  
sub—Use this keyword to identify the phones to be monitored.  
—Use this keyword to identify the specific users or extensions to be monitored.  
nme—(optional) Use this field with the SPA9000 to identify any alias that has been assigned to the  
extension in the Linksys IP phone configuration.  
vid—(optional) Use this field to identify the SPA962 extension to use with the monitored list  
specified by the sub= keyword, when more than one BLF monitored list is configured on the SIP  
proxy server. The possible values are 1 to 6, corresponding to each of the six extensions available  
on the SPA962.  
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Using the SPA932 (Sidecar) with the SPA962  
Note  
You only need to use the vid= keyword in the first entry assigned to each SPA962 extension.  
Subsequent keys will use the same extension.  
n with the extension on the SPA962 assigned to the LED on the SPA932  
The nme parameter indicates the extension name, which in this case is the same as the station name.  
Broadsoft syntax  
The following is an example entry for a Broadsoft server:  
fnc=enabled_functions;sub=blf_list_uri@broadsoft;usr=dn@broadsoft  
Replace the variables (in italics), as follows:  
enabled_functions with cp for call pickup, sd for speed dial, and blf for busy lamp field; separate  
multiple functions in any order with a plus sign (+)  
blf_list_uri with the URI assigned to the BLF monitoring list on the Broadsoft server.  
broadsoft with the IP address or hostname of the Broadsoft server.  
dn with the number associated with the user account on the Monitored Users list.  
For example, the following enables speed dialing and BLF monitoring for the user account reception,  
with a BLF monitoring list URI of marketing, on a Broadsoft server with the IP address 192.168.100.1:  
fnc=sd+blf;sub=marketing@;usr=reception@  
Note  
The nme keyword is not used because the Broadsoft server uses the user account name assigned  
to the BLF monitoring list.  
Asterisk syntax  
The following is an example entry for a Asterisk server:  
fnc=sd+blf+cp;sub[email protected];nme=35890  
This entry enables speed dialing and BLF monitoring on a Asterisk server with the IP address  
192.168.1.11:  
Step 10 Assign SPA932 LEDs to SPA962 extensions.  
By default, all 32 keys on the SPA932 are assigned to the first extension on the 962 (extension 101 based  
on the default multicast autoconfiguration).  
To limit the number of LEDs assigned to the first extension, use the vid=1 keyword. Then use the vid=2  
keyword to assign the next set of keys to the second extension. The numeric values correspond to the  
default extensions on the SPA962 as follows:  
vid=1—extension 101  
vid=2—extension 102  
vid=3—extension 103  
vid=4—extension 104  
vid=5—extension 105  
vid=6—extension 106  
Step 11 Repeat Step 9 for each extension or user that you are monitoring using the SPA932.  
Step 12 Click Submit All Changes at the bottom of the page.  
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Configuring the Web Service  
For instructions about how to configure phone extensions using the SPA Installation Wizard or the  
administration web server, refer to the SPA9000 Administration Guide.  
Step 13 Repeat Step 9 for each extension or user that you are monitoring using the SPA932.  
Step 14 Click Submit All Changes at the bottom of the page.  
Monitoring the SPA932  
To display the status of the SPA932, click the SPA932 Status link on the SPA932 tab. The sys-  
tem displays the screen shown in Figure 3-7.  
Figure 3-7 SPA932 Status  
For information about each field on this display, refer toChapter 5, “Linksys IP Phone Field Reference”.  
Configuring the Web Service  
Note  
Only web-enabled phones allow these features.  
Web Interface Basic and Advanced Views  
You can access configuration parameters using either the Basic or Advanced view. The SPA  
Provisioning tab is only visible from the Advanced Administrator view of the web interface.  
Warning  
Switching between the Basic and Advanced views discards the uncommitted changes that have been  
made on the web pages.  
Configuration Parameters  
You can configure SPA phone parameters in two ways:  
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Configuring the Web Service  
phone configuration web page  
remote provisioning  
In addition, a small subset of "user options" can be set via the phone’s GUI under various setup menus.  
On the configuration web page, the parameters are organized into several groups with each group listed  
under a tab on the top of the web page.  
To see a group of parameters, click on the corresponding tab on the web page. The following groups are  
defined:  
Info – Status Information (read only). This is the default tab when the web page is first loaded  
System – System level parameters, including network and debug parameters  
SIP – Parameters to adjust the SIP stack behavior  
Provisioning – Parameters that control remote provisioning  
Regional – Parameters that depends on country or region, such as call progress tones and * codes  
Phone – Parameters that apply to all the extensions configured for the phone  
Ext 1 – Parameters that apply to Extension 1  
Ext 2 – Parameters that apply to Extension 2  
Ext 3 – Parameters that apply to Extension 3  
Ext 4 – Parameters that apply to Extension 4  
User – User level parameters (other parameters are administrator level)  
Each parameter has a name and corresponding value. A parameter-name is of the form:  
seg[_seg][[n]]  
where seg is text segment composed of ascii characters only excluding spaces and underscores  
[ ]  
indicates optional components  
and n = 1, 2, 3, 4.  
Parameter names are case sensitive. Parameter values can be case sensitive or case insensitive,  
depending on the parameter. For example:  
Proxy[1], RTP_Packet_Size, DND_Act_Code  
When a parameter name is displayed on the web page, the underscore is replaced by a single space and  
the [n] component, if exists, is hidden. For example:  
Proxy, RTP Packet Size, DND Act Code  
Notes  
var (in italics) represents a variable or value to be substituted by the actual value it represents. var  
can appear standalone or anywhere within a longer string.  
a|b|c represents choices. The value can be one of the given choices.  
[optional value] represents an optional value  
<Par Name> represents a configuration parameter name, such as <Proxy>. In a profile, the  
corresponding tag is formed by replacing the space with an underscore "_", such as <RTP Packet  
Size> is replaced with RTP_Packet_Size. Some parameters carry an implicit index value if they  
apply to multiple extensions, such as <Proxy> which exists for Extensions 1, 2, 3, and 4. In a profile,  
"[n]" must be appended to the parameter to make it unambiguous. In this example, n = 1, 2, 3, 4 for  
Proxy[1], Proxy[2], Proxy[3], and Proxy[4] respectively.  
An empty default value field implies an empty string ("")  
The SPA continues to use the last configured values for tags that are not present in a given profile.  
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Configuring the Web Service  
Templates are compared in the order given. The first, not the closest, match is selected. The  
parameter name must match exactly.  
If more than one definition for a parameter is given in a configuration file, the the phone uses last  
definition in the file.  
A parameter specification with an empty parameter value forces the parameter back to its default  
value. To specify an empty string instead, use the empty string "" as the parameter value.  
Data Types  
Unsn – Unsigned n-bit value, where n = 8, 16, or 32. It can be specified in decimal or hex format  
such as 12 or 0x18 as long as the value can fit into n bits.  
Sign – Signed n-bit value. It can be specified in decimal or hex format. Negative values must be  
preceded by a "-" sign. A ‘+’ sign beforea positive value is optional.  
Int – A generic integer value. The range depends on the parameter.  
Strn– A generic string with up to n non-reserved characters.  
Floatn – A floating point value with up to n decimal places.  
Timen – Time duration in seconds, with up to n decimal places. Extra decimal places specified are  
ignored.  
PwrLevel – Power level expressed in dBm with 1 decimal place, such as –13.5 or 1.5 (dBm).  
Bool: Boolean value of either "yes" or "no".  
{a,b,c,…} – A choice among a, b, c, …  
IP – IP Address in the form of x.x.x.x, where x between 0 and 255. For example, 10.1.2.100.  
Port – TCP/UDP Port number (0-65535). It can be specified in decimal of hex format.  
UserID – User ID as appeared in a URL, up to 63 characters.  
FQDN – Fully Qualified Domain Name, such as "sip.linksys.com:5060", or "109.12.14.12:12345".  
It can contain up to 63 characters.  
Phone – A phone number string, such as 14081234567, *69, *72, 345678, or a generic URL such  
as [email protected]:5068, or [email protected]. It can contain up to 39 characters.  
ActCode – Activation code for a supplementary service, such as *69. It can contain up to 11  
characters.  
PhTmplt – A phone number template. Each template can contain 1 or more patterns separated by  
a ",". White space at the beginning of each pattern is ignored. The question mark "?" and asterisk  
"*" represent wildcard characters. The template can contain up to 39 characters. Examples: "1408*,  
1510*", "1408123????, 555?1".  
RscTmplt – A template of SIP Response Status Codes, such as "404, 5*", "61?", "407, 408, 487,  
481". It can contain up to 39 characters.  
CadScript – A mini-script that specifies the cadence parameters of a signal. Up to 127 characters.  
Syntax:  
1[;S2], where  
Si=Di(oni,1/offi,1[,oni,2/offi,2[,oni,3/offi,3[,oni,4/offi,4[,oni,5/offi,5[,oni,6/offi,6]]]]]) and is  
known as a section, oni,j and offi,j are the on/off duration in seconds of a segment and i = 1  
or 2, and j = 1 to 6.  
Di is the total duration of the section in seconds. All durations can have up to 3 decimal places  
to provide 1 ms resolution. The wildcard character "*" stands for infinite duration. The  
segments within a section are played in order and repeated until the total duration is played.  
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Configuring the Web Service  
Example 1: Normal Ring  
60(2/4)  
Number of Cadence Sections = 1  
Cadence Section 1: Section Length = 60 s  
Number of Segments = 1  
Segment 1: On=2s, Off=4s  
Total Ring Length = 60s  
Example 2: Distinctive Ring  
(short,short,short,long)  
60(.2/.2,.2/.2,.2/.2,1/4)  
Number of Cadence Sections = 1  
Cadence Section 1: Section Length = 60s  
Number of Segments = 4  
Segment 1: On=0.2s, Off=0.2s  
Segment 2: On=0.2s, Off=0.2s  
Segment 3: On=0.2s, Off=0.2s  
Segment 4: On=1.0s, Off=4.0s  
Total Ring Length=60s  
ToneScript – A mini-script that specifies the frequency, level and cadence parameters of a call  
progress tone. May contain up to 127 characters. Syntax:  
FreqScript;Z  
1[;Z2]. The section Zi is similar to the Si section in a CadScript except that each  
on/off segment is followed by a frequency components parameter: Zi =  
Di(oni,1/offi,1/fi,1[,oni,2/offi,2/fi,2 [,oni,3/offi,3/fi,3 [,oni,4/offi,4/fi,4 [,oni,5/offi,5/fi,5  
[,oni,6/offi,6/fi,6]]]]])  
where fi,j = n1[+n2]+n3[+n4[+n5[+n6]]]]] and 1 < nk < 6 indicates which of the frequency  
components given in the FreqScript are used in that segment; if more than one frequency  
component is used in a segment, the components are summed together.  
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Configuring the Web Service  
Example 1: Dial Tone  
350@-19,440@-19;10(*/0/1+2)  
Number of Frequencies = 2  
Frequency 1 = 350 Hz at –19 dBm  
Frequency 2 = 440 Hz at –19 dBm  
Number of Cadence Sections = 1  
Cadence Section 1: Section Length = 10 s  
Number of Segments = 1  
Segment 1: On=forever, with Frequencies 1 and 2  
Total Tone Length = 10s  
Example 2: Stutter Tone  
350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)  
Number of Frequencies = 2  
Frequency 1 = 350 Hz at –19 dBm  
Frequency 2 = 440 Hz at –19 dBm  
Number of Cadence Sections = 2  
Cadence Section 1: Section Length = 2s  
Number of Segments = 1  
Segment 1: On=0.1s, Off=0.1s with Frequencies 1  
and 2  
Cadence Section 2: Section Length = 10s  
Number of Segments = 1  
Segment 1: On=forever, with Frequencies 1 and 2  
Total Tone Length = 12s  
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Configuring the Web Service  
Example 3: SIT Tone  
985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,  
0/4/0)  
Number of Frequencies = 3  
Frequency 1 = 985 Hz at –16 dBm  
Frequency 2 = 1428 Hz at –16 dBm  
Frequency 3 = 1777 Hz at –16 dBm  
Number of Cadence Sections = 1  
Cadence Section 1: Section Length = 20s  
Number of Segments = 4  
Segment 1: On=0.38s, Off=0s, with Frequency 1  
Segment 2: On=0.38s, Off=0s, with Frequency 2  
Segment 3: On=0.38s, Off=0s, with Frequency 3  
Segment 4: On=0s, Off=4s, with no frequency components  
Total Tone Length = 20s  
RingScript – A script that describes a ring tone.  
Syntax: n=ring-tone-name;w=waveform-id-or-path;c=cadence-id;b=break-time;t=total-time  
ring-tone-name is a name to identify this ring tone specification. This name will appear on the  
Ring Tone menu of the phone. The same name can be used in a SIP Alert-Info header in an  
inbound INVITE request to tell the phone to play the corresponding ring tone specification.  
Because of this, the name should contain characters allowed in a URL only.  
Waveform-id is the index of the desired waveform to use in this ring tone specification. There  
are 4 built-in waveforms:  
1 = A classic phone with mechanical bell  
2 = Typical phone ring  
3 = A classic ring tone  
4 = A wide-band frequency sweep signal  
This field can also be a network path (url) to download a ring tone data file from a server  
on-the-fly. In this case, the syntax of the field is  
w=[tftp://]hostname[:port]/path.  
cadence-id is the index of the desired cadence to play the given waveform. 8 cadences (1–8) as  
defined in <Cadence 1> through <Cadence 8>. Cadence-id can be 0 If w=3,4, or an url. Setting  
c=0 implies the on-time is the natural length of the ring tone file.  
break-time specifies the number of seconds to break between two bursts of ring tone, such as  
b=2.5  
total-time specifies the total number of seconds to play the ring tone before it times out  
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RSS Newsfeeds (SPA962)  
Example 1: SIT Tone  
n=Classic-1,w=3;c=1  
n=Simple-1,w=2;c=1  
RSS Newsfeeds (SPA962)  
The SPA962 provides the option to view RSS newsfeeds. Press the Setup button-21 (Web Service) and  
select one of the following options on the LCD display:  
1. Stock Quote  
2. Traffic  
a. Option 2 shows "Traffic Service Information Settings" at the bottom.  
b. Enter your state, street, city and zip code to dispay local traffic information (defaults to CA,  
Cisco, San Jose, and 95134)  
3. Local News (defaults to url http://rss.cnn.com/rss/cnn_us.rss)  
4. World News (defaults to url http://newsrss.bbc.co.uk/rss/newsonline_uk_edition/world/rss.xml)  
5. Finance News (defaults to http://finance.yahoo.com/rss/topstories)  
6. Sport News (defaults to http://rss.news.yahoo.com/rss/sports)  
7. Politics News (defaults to http://rss.news.yahoo.com/rss/politics)  
Option 3 to 7 are configurable using the administration web server, as shown in the following procedure.  
Step 1  
Step 2  
Step 3  
Step 4  
Connect to the administration web server for the SPA962.  
Click Admin/Advanced on the administration web server page.  
Click the Phone tab.  
Change the name or URL used for each of the configurable options:  
Option 3—RSS Feed URL 1:  
Option 4—RSS Feed URL 2:  
Option 5—RSS Feed URL 3:  
Option 6—RSS Feed URL 4:  
Option 7—RSS Feed URL 5:  
Step 5  
Step 6  
Change the update interval, if desired (defaults to 15)..  
Change the default scrolling timer, if desired (defaults to 0.8).  
When the news information is too large to be displayed in one window, the display automatically scrolls.  
The Scrolling Timer defines the speed of the scrolling; the default is .8 seconds.  
Step 7  
Click Submit All Changes at the bottom of the page.  
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Call Appearances and Extensions  
Call Appearances and Extensions  
A client station is a Linksys IP phone with one or more extensions and call appearances provisioned An  
extension in the Linksys IP phone is a VoIP 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).  
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 Linksys IP 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 on hold.  
The VoIP Interface (VI) refers to an extension and its associated control parameters that are configured  
on a particular station. Linksys IP phones include a rich set of configurationx parameters to control the  
operation of the phone. Configuring an extension includes configuring the core account information and  
VI control parameters.  
Although 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  
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  
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 another station (shared line 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  
state of the Call is Active, it is referred to as an Active Call  
Standby (Call) State: the state is Ringing or Holding. When the state of the Call is Standby, it is  
referred to as a Standby Call  
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Line Key LEDs  
Line Key LEDs  
This section summarizes the behavior of the line key LEDS on Linksys IP phones with multiple line key  
status indicators. For more information, see “Line Key LED Pattern” section on page 5-31.  
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 the LedScript.  
LED Script  
The LED script describes the color and blinking pattern of a Line Key LED. Each script contains a  
number of fields separated by a semicolon(;). White spaces are ignored. Each field has the syntax  
<field-name> = <field-value>. The allowed field-name and corresponding field-values are listed below:  
c=o|r|g|a  
This field sets the color of the LED. The 4 choices are:  
o = off  
r = red  
g = green  
a = amber  
p=n[b]|s[b]|f[b]|d[b]|u[d]  
This field sets the blinking pattern of the LED. The 4 choices are:  
nb = no blink (steady on or off)  
sb = slow blink (1s on and 1s off)  
fb = fast blink (100ms on and 100ms off)  
ud = user-defined (according to the contents of the u field)  
u=on/off/on/off...  
This is a user-defined blinking pattern used only when p = ud. It consists of up to 4 pairs of on/off  
duration in seconds with up to 2 decimal places; each value is separated by a forward slash (/).  
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Line Key LEDs  
LED Script Examples  
Example 1  
c=r;p=sb  
Color is red and slow blink  
Example 2  
c=o  
LED is off  
Example 3  
c=g  
Color is green and steady on  
Example 4  
c=a;p=ud;u=.1/.1/.1/.1/.1/.9  
Color is amber and blink with the pattern (100ms on, 100ms off, 100ms on, 100ms off,  
100ms on, 900ms off).  
LED Pattern  
The administrator can also specify a different color and pattern for each of the following states of the  
call appearance.  
Idle: This call appearance is not in use. It can be used 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  
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  
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  
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Using Call Features  
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  
The Linksys IP phones provide three audio I/O devices:  
Handset  
Built-in microphone and speaker  
External microphone and headset  
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 one of the calls becomes active,  
the handset is selected automatically as the audio device if it is off-hook; otherwise, the speaker or the  
headset is selected according to configuration. You can configure whether the speaker or the headset  
device has higher preference.]  
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Using Call Features  
A line is selected by pressing the corresponding line key. To make or receive calls, you must select a call  
appearance (line key) or an audio device. If you select a call appearance, then an audio device is selected  
automatically according to user preference setting. If you select an audio device, an idle call appearance  
is selected automatically in the order L1, L2, L3, and L4. You can 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 (Conf on the SPA962)  
or xfer (Transfer on the SPA962) 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).  
Making Calls  
There are two steps in making a call:  
1. Select an audio device  
2. Dials the number  
The Linksys IP phone lets you perform these two steps in either order. In other words, the user can select  
an audio device either before or after dialing. The Linksys IP phone supports two types of dialing:  
explicit or implicit. Explicit dialing occurs when the user performs one of the following actions:  
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  
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 ascending 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. You 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.  
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.  
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Using Call Features  
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. To 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 Waiting function is activated when a device has a call in the active state and another call is  
incoming. Linksys IP phones 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 you, the call waiting tone is played into 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.  
Speed Dialing  
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 Setup 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. You 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. 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. (You 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 you to select the other line key to join with the conference call.  
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Using Call Features  
Once the conference starts, the 900 Series phone plays a special brief tone to all three 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 can 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:  
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  
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 activate 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, you are 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.  
Blind Call Transfer  
A blind call transfer allows you 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 transferred.  
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Using Call Features  
Call Back  
This service mimics the call back service offered by the PSTN. You can activate this service on a busy  
number so that you are 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, you select a line key and enters *66 (call back activation code). The 900 Series  
phone then uses this call appearance to call the last called number. The retry period and expiration time  
are configurable. To cancel the service, you pick up any line and enter *86. When call back service is  
active, the corresponding Line key flashes green. You 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, you hear a ring back tone like a regular  
outbound call.  
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 Series phone can be configured to play a stutter tone on the individual call appearance according to  
the message waiting state of the corresponding Extension X.  
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 you 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  
You 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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Using Call Features  
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  
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. To view or edit the personal directory via the  
phone web interface, use the link http://ip_address/pdir.htm.  
You can invoke the Directory menu on the phone in 2 ways:  
Press the Setup button and select Directory  
Press dir when not already in a Setup display  
You 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 you for confirmation before  
proceeding.  
You can copy an existing directory entry to create a new entry. To do this, highlight the entry you want  
to copy and press copy. Then highlight the New Entry option and press paste. You can edit the new entry  
and then press save to save the entry. You 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 match 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 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.  
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Using Call Features  
Dialing Assistance  
Dialing assistance is an option you 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 you dial 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, you 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  
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. You can also enable or disable this feature on a per call basis by pre dialing 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. You 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  
There are three call logs maintained for each VoIP interface:  
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 chronological 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  
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Using Call Features  
Each log is shown on the web page with the following syntax:  
name,ext-id,date-time stamp  
Where:  
ext-id is the extension or phone number of the call peer.  
name is the call peer name. If this field is missing, the peer name is not available.  
date-time stamp is the time at which the call log is entered. It has the format mm/dd hh/mm<a|p> (a  
= am and p = pm).  
For example:  
Joe Smith,14089991234,10/18 11:15a  
No name appears if the name is not available.  
Table 3-4 summarizes call log operations..  
Table 3-4 Call Log Operations  
Operation  
Steps  
To view a call log  
1. Press Menu.  
2. Select Call History.  
3. Select the log you want to view.  
1. Highlight the entry.  
2. Press del.  
To delete a log  
To edit a log  
1. Highlight the entry.  
2. Press edit.  
3. Press save to save the changes, or cancel to abort the change.  
1. Highlight the entry.  
To call a log entry  
2. Press dial, or Off-Hook, or Turn on Speaker, or Turn on  
Headset, or press a line key.  
To save a log entry into Personal 1. Highlight the entry.  
Directory (if the directory is not  
already full)  
2. Modify the entry as needed.  
3. Press save to save the entry in the personal directory.  
Audio Volume Adjustment  
You can adjust four kinds of audio volume. Note that you must save the volume immediately after  
adjusting:  
Ringer volume (on the speaker)  
Speaker  
Handset  
Headset  
To adjust the volume, press the VOL key to invoke the volume adjustment screen. Use the UP/DOWN  
keys to increase or decrease the volume, then press the Save softkey. The kind of volume to adjust is by  
context. When no audio device is selected at the moment, the Ringer volume is adjusted. Each  
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Using Call Features  
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, you can assign a name for the ring tone, and specify two  
components:  
Waveform—1, 2, 3, or 4  
Cadence—1, 2, 3, or 8  
You can assign a default ring tone for each extension. You 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.  
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, obtain a .wav file less than 8 seconds in length and convert it using  
the Upload Ringtone utility available from the following website:  
Scroll down to SPA Firmware with Upgrade Utility Program > SPA-841, Version 3.0 and click on  
Ringtone_Generator.  
Figure 3-8 illustrates the user interface provided by this utility.  
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Chapter 3 Managing Linksys IP Phones  
Configuring a Dial Plan  
Figure 3-8 Upload Ringtone Utility  
For input, you can use any .wav file less than eight seconds.  
Configuring a Dial Plan  
The Linksys IP 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 Linksys IP phone syntax for the dial plan closely resembles the corresponding 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 Linksys IP 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 number 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.  
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.  
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Configuring a Dial Plan  
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.  
If you are using the SPA9000, enter 9 > *69 to callback and call return.  
Table 3-5 describes the entries to use when programming the dial plan.  
.
Table 3-5 Dial Plan Entries  
Dial Plan Entry  
*xx  
Function  
Allows arbitrary 2-digit star code  
Allows x11 sequences (for example, 311, 411, 611, 911)  
Dials operator  
[3469]11  
0
00  
Dials international operator  
[2-9]xxxxxx  
1xxx[2-9]xxxxxx  
xxxxxxxxxx.  
Dials US local number  
Dials US 1 + 10-digit long distance number  
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): for example, [389] means 3 or 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  
Any element can be repeated zero or more times by appending a period (.) to the element. Thus, “01.”  
matches “0”, “01”, “011”, “0111”, … and so on.  
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”.  
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Configuring a Dial Plan  
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.  
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.  
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.  
Implicit Sequences  
The Linksys IP 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 )  
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Configuring a Dial Plan  
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 )  
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 timeout to 4 seconds.  
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 )  
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  
Interdigit_Long_Timer  
10  
ParName  
Default  
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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System Administration  
Interdigit Short Timer  
Interdigit_Short_Timer  
3
ParName  
Default  
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  
Dial_Plan[n] for Each Line n  
ParName  
Default  
(*xx | [3469]11 | 0 | 00 | <:1408>[2-9]xxxxxx |  
1[2-9]xx[2-9]xxxxxx | 011x. )  
The Dial_Plan parameters contain the actual dial plan scripts for each line n, where n is a number from  
1 to 4.  
System Administration  
This section describes features and operations required for administering and maintaining the basic  
function of the Linksys IP phone. It includes the following topics:  
Reboot and Restart  
Reboot cold boots the phone and restarts the phone’s DHCP client.  
Restart warm boots the phone, without restarting the phone’s DHCP client.  
You must confirm the chosen option before continuing.  
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System Administration  
Factory Reset  
Factory reset resets all parameters to default value. The personal directory and call logs are also cleared.  
You must confirm this option before continuing.  
Password Protection  
You can configure a password on the phone to protect against access to the certain options.  
Managing the Time/Date  
The Linksys IP phone obtains the current time information in one of the following ways:  
NTP Server—You can configure one or two NTP servers on the phone. When the phone first boots  
up, it tries to contact the first NTP server to get the current time. The phone periodically  
synchronizes its time with the NTP server. The synchronization period is fixed at 1 hour. In between  
updates, the phone tracks 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.  
Manual Setup—The Linksys IP phone also lets you manually enter the current time and date from  
the phone GUI or from the web page. However, this value is overridden by the NTP time or SIP  
Message Date whenever they are presented to the phone. Manual setup requires you to enter the time  
in 24 hour format only.  
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 Offset (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 Linksys IP phone supports auto adjustment for daylight saving time. To enable this feature, the  
administrator must configure the <Daylight Saving Time 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]]] The <save-time> value can be preceded by a negative  
(-) sign if subtraction is desired instead of addition.  
<month> = 1,2,12 (for Jan, Feb, Dec)  
<day> = [+|-]1,2,3,...,,31. 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).  
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System Administration  
<weekday> = 1, 2, 3,..., 7 (for Mon, Tue, ..., Sun), or 0. 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.  
HH = hour (0-23)  
mm = minute (0-59)  
ss = second (0-59)  
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 ; add 1 hour  
(USA, North America):  
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  
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:  
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. This only applie to  
countries that recognize daylight savings time.  
start=10/1/7;3/22/7;save=1 (New Zealand)  
The following example reflects the new change starting March 2007. DST starts on the second Sunday  
in March and ends on the first Sunday in November:  
start=3/8/7/02:0:0;end=11/1/7/02:0:0;save=1  
Using Star Codes to Activate/Deactivate Services  
The Linksys IP phone accepts star codes to activate or deactivate certain services on the phone.  
To activate a star code, either:  
Off hook the phone and dial the * code (if you know the star code number)  
or  
Off hook the phone, press the right arrow to see/ press the button for star codes, scroll through the star  
code list and select the desired star code (if you do not know the star code number)  
Below is a list of services that are accessible with a star code, with the default star code shown in  
parentheses.  
Call Return (*69)—Calls the last caller, regardless which extension.  
Call Back(*66)—Periodically redials the last busy number (every 30 seconds by default) until it  
rings or until the attempt expires (30 min by default), regardless which extension. Only one 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.  
Call Forward All (*72)—Call forwards all inbound calls. Applies to primary extension only  
Cancel Call Forward All (*73)—Cancels call forward all. Applies to primary extension only  
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System Administration  
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 outbound call (on the current call  
appearance only).  
Secure All Calls (*16)—Defaults to prefer to use encrypted media (voice codecs).  
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.  
Cancel Do Not Disturb (*79)—Applies to all extensions.  
Block Anonymous Calls—Applies to all extensions.  
Cancel Block Anonymous Calls—Applies to all extensions.  
Blind Transfer (*98)—This allows you to transfer a call to another person without talking to that  
person:  
a. Put the call on hold  
b. Get a dial tone on another line  
c. Call the person to whom you are transferring the call by pressing *98, followed by their  
extension  
d. Hang up the receiver  
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.  
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System Administration  
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, you 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: A list of star codes separated by a vertical bar (|), such as *199|*09, where  
entering 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: 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. May be useful for some Service Providers who have their own star codes. These Service  
Provider star codes can be listed under the Feature Dial Services Codes.  
Disabling Services  
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.  
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 Linksys IP 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.  
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.  
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Troubleshooting FAQ  
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 message waiting light LED has the following Blink Rate Definitions:  
ON—LED remains solid on (indicates voicemail, no errors)  
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  
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 partner PIN)  
888 333-0244 Hours: 4am-6pm PST, 7 days a week  
E-mail support  
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.  
Q. Why do the line status lights on my Linksys IP phone occasionally flash red?  
A. A flashing red line status light on the Linksys IP phone indicates a local LAN issue. If statistics on  
the switch show a high number of collisions on the ports connected to the Linksys IP phone, you  
can reduce the speed on the ports to 10mb half duplex to eliminate the collision problems.  
Q. Why can I not leave a voicemail message longer than 30 seconds?  
A. If you are using 1.0.0.3 firmware, upgrade SPA400 to version 1.0.0.5 or greater. The upgrade will  
let you leave a message of 60 seconds.  
Q. Why does the phone does not turn on or boot?  
A. Check all connections into the phone (5 Volt Power supply, RJ 45 Ethernet connection, and handset  
cord).  
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Troubleshooting FAQ  
If using PoE, verify that the RJ-45 connector is firmly connected to the switch/wall connection.  
If not using PoE, check the wall socket and confirm that AC power is available. Also, try  
plugging another electrical device into the socket to confirm power.  
Q. Why does the phone not make or receive calls?  
A. If the phone does not make or receive calls, first verify that the phone’s line LEDs are green. Any  
other color indicates either a registration or network problem. If the LEDs are not green:  
1. Press the Setup 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  
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 Setup button.  
4. Get the current IP address of the phone 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.  
6. Get the current IP address of the phone 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 address and registration status (Registered,  
Not Registered).  
Q. Why are voice calls of poor sound quality?  
A. 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)  
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C H A P T E R  
4
SPA900 Series LCD Command Reference  
This chapter provides a systematic reference for the configuration and monitoring options available from  
the LCD display on 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 “Using the  
This chapter includes the following sections:  
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1 Directory  
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:  
Note  
The following sections show the soft key names for the SPA962 in parentheses ( ).  
Entering Names and Numbers into the Directory  
Shortcut: Setup, 1, add, name, scroll down, number, scroll down or save  
To enter the Directory option, press the Setup button and then press the 1 button. Alternatively, you  
may press the Setup button and the Directory option is highlighted. Then press the select soft key. After  
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.  
Entering Directory Names, Numbers and Ring Default  
After pressing add, you are prompted to enter a Name. To enter a letter of a name, press the 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 down. 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. Press clear to clear the whole string.  
To enter a phone number into the directory, simply dial the number including the 1 and the Area Code  
(for example, 14085721111).  
Note  
Enter 9 first to call an outside line.  
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 for 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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2 Speed Dial  
Note  
Be sure to press the save button after entering information. Otherwise, the information is not saved and  
will not appear in the directory.  
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.  
To set-up a speed dial entry, press the Setup button and then press 2. Alternatively, press the Setup  
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 the number (or scroll down until it is highlighted), then 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 address 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 you enter letters, the phone displays names in the  
directory that match the letters displayed. Use the scroll buttons 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.  
Note  
To assign a new name to an already assigned number, perform the same steps but clear the old entry  
before entering the new name.  
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 #.  
3 Call History  
Shortcut:: Setup, 3  
To access Call History, press the Setup button, then press 3. Alternatively, you can press Setup, scroll  
down until call history is highlighted, and then press the select soft key.  
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 (right soft  
key).  
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4 Ring Tone  
Redial List  
Shortcut:: Setup, 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 Setup option, press the cancel soft key.  
Answered Calls  
Shortcut:: Setup, 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 Setup option, press the cancel soft key.  
Missed Calls  
Shortcut:: Setup, 3, 3  
After selecting this option, the phone displays the call entries in the Missed Calls 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 Setup option, press the cancel soft key.  
4 Ring Tone  
Shortcut:: Setup, 4, select extension, change, select ring tone  
This option lets you select a different ring tone for each extension that the phone supports. Phones can  
have distinctive ring tones, which can be useful in a shared office environment.  
To set or change the ring tone for a specific extension, press Setup and then select ring tones by pressing  
4. Alternatively, press the Setup button and scroll down to the ring tone 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.  
To select a ring tone, press the corresponding number or highlight the option. 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:  
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5 Preferences  
5.1 Block Caller ID  
5.2 Block Anonymous callers  
5.3 Do Not Disturb  
5.4 Secure Call  
5.5 Dial Assistance  
5.6 Preferred Audio Device  
7 Auto Answer Page  
8 Send Audio To Speaker  
9 Time Format  
10 Date Format  
11 Miss Call Shortcut  
12 Call Waiting  
To select the Preferences option, press the Setup 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 down to the selection and press the 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: Setup, 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 the Setup button and then press 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: Setup, 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 Setup 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.  
5.3 Do Not Disturb  
Shortcut: Setup, 5, 3, edit, y/n, ok  
Do Not Disturb routes the call to voicemail or presents a busy signal if voicemail is not available.  
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5 Preferences  
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.  
5.4 Secure Call  
Shortcut: Setup, 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 Linksys IP 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.  
5.5 Dial Assistance  
Shortcut: Setup, 5, 5, edit, y/n, ok  
The Dial Assistance option allows the Linksys IP phone to automatically suggest a number 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: Setup, 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. With the Preferred Audio  
Device showing the desired selection, press ok. The display returns to the Preferences menu.  
5.7 Auto Answer Page  
Shortcut: Setup, 5, 7, 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. With the Preferred Audio  
Device showing the desired selection, press ok. The display returns to the Preferences menu.  
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5 Preferences  
5.8 Preferred Audio Device  
Shortcut: Setup, 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. With the Preferred Audio  
Device showing the desired selection, press ok. The display returns to the Preferences menu.  
5.9 Preferred Audio Device  
Shortcut: Setup, 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. With the Preferred Audio  
Device showing the desired selection, press ok. The display returns to the Preferences menu.  
5.10 Preferred Audio Device  
Shortcut: Setup, 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. With the Preferred Audio  
Device showing the desired selection, press ok. The display returns to the Preferences menu.  
5.11 Preferred Audio Device  
Shortcut: Setup, 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. With the Preferred Audio  
Device showing the desired selection, press ok. The display returns to the Preferences menu.  
5.12 Preferred Audio Device  
Shortcut: Setup, 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. With the Preferred Audio  
Device showing the desired selection, press ok. The display returns to the Preferences menu.  
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6 Call Forward  
6 Call Forward  
To view or change the call forwarding setting, press the Setup 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: Setup, 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  
Shortcut: Setup, 6, 2, edit, phone_number, ok  
This option forwards calls to another number when the phone is busy.  
To forward all calls to specific telephone number when the phone is busy Press Setup, 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: Setup, 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: Setup, 6, 4, edit, phone_number, ok  
This option sets the time interval after which forwarding occurs.  
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7 Time/Date  
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: Setup, 7 ,date, time, offset, save  
To set the date and time, press Setup 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: Setup, 8, voicemail_number, save  
To set the number to call voicemail, press Setup and then press 8, or scroll down until the selection is  
highlighted and press the select soft key.  
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 Setup button) to dial the mailbox.  
9 Network  
To view or modify network parameters, press the Setup button and then press 9, or scroll down until  
Network is highlighted and then press the select soft key. This menu displays the IP network  
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  
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9 Network  
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  
9.15 Multicast Address  
9.16 Enable VLAN  
9.17 VLAN ID  
9.18 CDP  
9.1 DCHP  
Shortcut: Setup, 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 Setupbutton 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  
Shortcut: Setup, 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:  
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  
9.4 Domain  
Shortcut: Setup, 9, 3 save  
The Host Name is set by the administrator. 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.  
Shortcut: Setup, 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 scroll down to highlight Domain  
and press select.  
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9 Network  
9.5 Current NetMask  
Shortcut: Setup, 9, 5 save  
To 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 following 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: Setup, 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 (you cannot edit it)  
Manually set the address using the Non DHCP Default Route option (Option 9.10)  
9.7 Enable Web Server  
Shortcut: Setup, 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.  
9.8 Non DHCP IP Address  
Shortcut: Setup, 9, 8, ip_address, ok  
To manually set the IP address, press 8 from the Network menu, or scroll down to highlight 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: Setup, 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 subnet  
mask, press ok.  
9.10 Non DHCP Default Route  
Shortcut: Setup, 9, 10, ip_address, ok  
To edit the default gateway (router), press 10 or scroll down to highlight Non DHCP Default Route and  
press the select soft key. Then press the edit soft key. After entering the default route, press ok..  
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9 Network  
9.11 Non DHCP DNS 1  
Shortcut: Setup, 9, 11, ip_address, ok  
To edit the primary 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: Setup, 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: Setup, 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.  
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 using 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: Setup, 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.  
9.15 Multicast Address  
Shortcut: Setup, 9, 15, multicast_address, ok  
To edit the secondary NTP server used by the phone, press 15 from the Network menu, or scroll down  
and highlight Multicast Address and press the select soft key. Then press the edit soft key. After  
entering the address, press ok.  
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10 Product Info  
9.16 Enable VLAN  
Shortcut: Setup, 9, 16, y/n, ok  
To enable VLAN, press 16 from the Network menu, or scroll down and highlight Enable VLAN and  
press the select soft key. Then 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.  
9.17 VLAN ID  
9.18 CDP  
Shortcut: Setup, 9, 17, VLAN_ID, ok  
To edit the VLAN ID, press 17 from the Network menu, or scroll down and highlight VLAN ID and  
press the select soft key. Then press the edit soft key. After entering the ID, press ok.  
Shortcut: Setup, 9, 18, y/n, ok  
To enable Cisco Discovery Protocol (CDP), which allows Cisco devices to discover one another  
automatically, press 18 from the Network menu, or scroll down and highlight CDP and press the select  
soft key. Then 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.  
10 Product Info  
To view the parameters in the Product Info, press the Setup 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  
10.2 Serial Number  
10.3 Software Version  
10.4 Hardware Version  
10.5 MAC Address  
10.6 Client Cert  
10.7 Customization  
10.1 Product Name  
Shortcut: Setup, 10, select  
To view the Product Name, press the Setup 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.  
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11 Status  
10.2 Serial Number  
Shortcut: Setup, 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  
Shortcut: Setup, 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.  
10.4 Hardware Version  
Shortcut: Setup, 10, 4, select  
To 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: Setup, 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: Setup, 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.  
10.7 Customization  
Shortcut: Setup, 10, 7, select  
The phone can be configured for customization. Press 7 from the Product Info menu, or scroll down to  
the highlight Customization, and press the select soft key. “Open” indicates that the phone can be  
customized.  
11 Status  
To view the status of the phone and extension parameters, press the Setup button 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.  
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11 Status  
Note  
On the SPA962 phone, “EXT” appears as “Ext,” and “Line” appears as “Line Key.”  
Phone  
EXT 1  
EXT 2  
EXT 3  
EXT 4  
Line 1  
Line 2  
Line 3  
Line 4  
Phone  
Shortcut: Setup, 11, select  
To view the phone status, press the Setup button and then 11 or scroll down to Status and press the select  
soft key. The display provides the date and time, Up time, Transmitted packets (Tx) and Received  
Packets (Rx).  
Ext 1/2/3/4  
Shortcut: Setup, 11, 2 or 3 or 4 or 5 select  
To 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 time and date, and the  
time in seconds until the next registration (Next Reg). If the extension is not registered, the phone  
displays Not Registered along with the proxy’s IP address and port number.  
Line 1, 2,3,4  
Shortcut: Setup, 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.  
Provisioning  
Shortcut: Setup, 11, 14 select  
To see the provisioning status for your phone, from the Status menu, scroll to 14, then press select. The  
display shows the profile, current provisioning status (Failed or Successful). If Failed, the reason for the  
failure is displayed.  
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12 Reboot  
Call Statistics History  
Shortcut: Setup, 11, 15 select  
To see the call statistics history for your phone, from the Status menu, scroll to 15, then press select. The  
display shows the call history. If there is no call history logged, the phone displays “No Item.”  
12 Reboot  
13 Restart  
Shortcut: Setup, 12, ok  
This option lets you reboot the phone. To select Reboot, press the Setup button and press 12, or scroll  
down until the selection is highlighted and then press select soft key. The LCD displays Confirm  
System Reboot? You can choose either ok (OK on the SPA962) or Cancel. Select ok (OK on the  
SPA962) to reboot the phone. The option is only displayed for 5 seconds. After 5 seconds, the display  
returns to the Setup option.  
Shortcut: Setup, 13, ok  
This option lets you restart the phone. To select Restart, press the Setup 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? You can choose either ok (OK on the SPA962) or Cancel.  
Select ok (OK on the SPA962) to restart the phone. This option is only displayed for 5 seconds. After 5  
seconds, the display is returned to the Setup option.  
14 Factory Reset  
Shortcut: Setup, 14, ok  
This option lets you reset the parameters to the factory settings. To select Factory Reset, press the Setup  
button and press 14, or scroll down until the selection is highlighted and press the select soft key.  
The menu displays All parameters will be reset to factory default values. Continue?  
You can choose either ok (OK on the SPA962) or Cancel. Select ok (OK on the SPA962) to 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 Setup option.  
15 Custom Reset  
Shortcut: Setup, 15, ok  
The Custom Reset feature works with the Remote Customization (RC) unit. RC units have different  
configurations, depending on the ITSP. Custom Reset allows a Service Provider to change its original  
configuration. This is a one-time install; after the RC is configured for a specific Service Provider, it  
cannot be changed.  
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Chapter 4 SPA900 Series LCD Command Reference  
16 Set Password  
When a user receives an RC unit, the unit obtains the correct profile for its ITSP using the MAC and  
server addresses.  
16 Set Password  
Shortcut:: Setup, 16, password, ok  
This option lets you set a password to protect the phone from unauthorized use. To select this option,  
press the Setup button and press 15, or scroll down until the selection is highlighted and press the 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.  
17 Set LCD Contrast  
Shortcut:: Setup, 17, scroll knob left or right, save  
This option lets you adjust the LCD contrast. To use this option, press the Setup button and press 16  
scroll down to highlight this option and press the select soft key.  
Press the Navigation button right or left to adjust the LCD contrast up or down. Press the save soft key  
when finished, or the cancel soft key to keep the previously selected LCD contrast.  
18 CallPark Status  
Shortcut: Setup, 18  
This option lets you display the status of the CallPark feature. When this option is selected, the LCD  
displays the Requesting CallPark Status message.  
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.  
19 Language (SPA922, 942, and 962)  
Shortcut:: Setup, 19, 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 you change the language of the phone:  
1. Press the Setup button.  
2. Select Language (option 18).  
3. Select Edit to change language.  
4. Select Option to change the language.  
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Chapter 4 SPA900 Series LCD Command Reference  
20 GUI Properties (SPA962)  
5. Select OK and Save.  
Dictionaries are currently available for the following languages:  
English  
Spanish  
German,  
Dutch  
Italian  
French  
Portuguese  
Danish  
Swedish  
Czech  
Slovak  
For information about configuring the languages available through the LCD menu, refer to  
20 GUI Properties (SPA962)  
The SPA962 lets you change the background on the LCD display. You can change:  
Preferred Background Picture: BMP Picture, Default, or None  
Preferred Menu Color Scheme: Light Blue, Light Green, Pink, Silver  
21 Web Service (SPA962)  
The SPA962 provides the option to view RSS newsfeeds by selecting one of the following options on  
the LCD display:  
1. Stock Quote.  
2. Traffic  
3. Local News  
4. World News  
5. Finance News  
6. Sport News  
7. Politics News  
Option 3 to 7 are configurable using the administration web server, as described in “Configuring the Web  
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C H A P T E R  
5
Linksys IP Phone Field Reference  
This chapter describes the fields within each section of the following administration web server pages:  
For information about the Provisioning page, see the Linksys SPA Provisioning Guide.  
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Info Tab  
Info Tab  
This section describes the fields for the following headings on the Info tab:  
Note  
The fields on this tab are read-only and cannot be edited.  
System Information  
Field  
Description  
DHCP  
Indicates if DHCP is enabled. Displays Enabled on SPA901, DHCP on other phones.  
Displays the current IP address assigned to the Linksys IP phone.  
Displays the current host name assigned to the Linksys IP phone (defaults to SipuraSPA).  
Displays the network domain name of the Linksys IP phone.  
Current IP  
Host Name  
Domain  
Current Netmask  
Current Gateway  
Primary DNS  
Secondary DNS  
Displays the network mask assigned to the Linksys IP phone.  
Displays the default router assigned to the Linksys IP phone.  
Displays the primary DNS server assigned to the Linksys IP phone.  
Displays the secondary DNS server assigned to the Linksys IP phone.  
Product Information  
Field  
Description  
Product Name  
Serial Number  
Software Version  
Hardware Version  
MAC Address  
Client Certificate  
Model number of the Linksys IP phone.  
Serial number of the Linksys IP phone.  
Version number of the Linksys IP phone software.  
Version number of the Linksys IP phone hardware.  
MAC address of the Linksys IP phone.  
Status of the client certificate, which authenticates the Linksys IP phone for use in the ITSP  
network. This field indicates if the client certificate is properly installed in the Linksys IP  
phone.  
Customization (not applicable  
to SPA901)  
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 additional licenses that you have installed in the Linksys IP phone.  
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Chapter 5 Linksys IP Phone Field Reference  
Info Tab  
Phone Status  
Field  
Description  
Current date and time of the system; for example, 10/3/2003 16:43:00.  
Current Time  
Elapsed Time  
Total time elapsed since the last reboot of the system; for example, 25 days and 18:12:36.  
Total number of broadcast packets sent.  
Broadcast Pkts Sent  
Broadcast Bytes Sent  
Broadcast Pkts Recv  
Broadcast Bytes Recv  
Broadcast Pkts Dropped  
Broadcast Bytes Dropped  
RTP Packets Sent  
RTP Bytes Sent  
Total number of broadcast packets received.  
Total number of broadcast bytes sent.  
Total number of broadcast bytes received and processed.  
Total number of broadcast packets received but not processed.  
Total number of broadcast bytes received but not processed.  
Total number of RTP packets sent (including redundant packets).  
Total number of RTP packets received (including redundant packets).  
Total number of RTP bytes sent.  
RTP Packets Recv  
RTP Bytes Recv  
SIP Messages Sent  
SIP Bytes Sent  
Total number of RTP bytes received.  
Total number of SIP messages sent (including retransmissions).  
Total number of SIP messages received (including retransmissions).  
Total number of bytes of SIP messages sent (including retransmissions).  
Total number of bytes of SIP messages received (including retransmissions).  
External IP address used for NAT mapping.  
SIP Messages Recv  
SIP Bytes Recv  
External IP  
Operational VLAN ID  
ID of the VLAN currently in use.  
(not applicable to SPA901)  
Ext 1/2/3/4/5/6 Status  
)
Field  
Description  
Registration State  
Shows “Registered” if the phone is registered, “Not Registered” if the phone is not  
registered.  
Last Registration At  
Last date and time the line was registered.  
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Chapter 5 Linksys IP Phone Field Reference  
Info Tab  
Field  
Description  
Next Registration In  
Message Waiting  
Number of seconds before the next registration renewal.  
Indicates whether you have new voicemail waiting: Yes or No. This is updated when  
voicemail notification is received.  
Mapped SIP Port  
Port number of the SIP port mapped by NAT.  
Line 1/2/3/4/5/6 Call 1/2 Status  
Field  
Description  
Hazardous Potential  
Foreign Voltage  
The amount of hazardous potential is displayed.  
The amount of foreign voltage is displayed.  
Long Resistive Fault  
REN  
The amount of long resistive fault is displayed.  
The Ringer Equivalence Number (REN) is displayed.  
Call State  
Tone  
Status of the call.  
Type of tone used by the call.  
Codec used for encoding.  
Encoder  
Decoder  
Codec used for decoding.  
Type  
Direction of the call.  
Remote Hold  
Callback  
Indicates whether the far end has placed the call on hold.  
Indicates whether the call was triggered by a call back request.  
Name of the internal phone.  
Peer Name  
Peer Phone  
Duration  
Phone number of the internal phone.  
Duration of the call.  
Packets Sent  
Packets Recv  
Bytes Sent  
Bytes Recv  
Decode Latency  
Jitter  
Number of packets sent.  
Number of packets received.  
Number of bytes sent.  
Number of bytes received.  
Number of milliseconds for decoder latency.  
Number of milliseconds for receiver jitter.  
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Info Tab  
Round Trip Delay  
Packets Lost  
Number of milliseconds for delay.  
Number of packets lost.  
Packet Error  
Number of invalid packets received.  
Mapped RTP Port  
The port mapped for Real Time Protocol traffic for the call.  
Media Loopback  
(not applicable to SPA901)  
If the call is a loopback call, displays the loopback mode (source or mirror) and type  
(media or packet). If the call is not loopback, the field appears blank.  
Downloaded Ring Tone  
Field  
Description  
Status  
Indicates whether the phone is downloading a ring tone (and from where) or if it is idle.  
Ring Tone 1  
Information about the user downloaded ring tone 1: name, size, and time-stamp of the  
tone.  
Ring Tone 2  
Information about the user downloaded ring tone 2: name, size, and time-stamp of the  
tone.  
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System Tab  
System Tab  
This section describes the fields for the following headings on the System tab:  
System Configuration  
Field  
Description  
Restricted Access Domains  
(not applicable to SPA901)  
This feature is used when implementing software customization.  
Enable Web Server  
(not applicable to SPA901)  
Enable/disable web server of Linksys IP phone  
This feature should only be used on firmware version 1.0.9 or later.  
Defaults to yes.  
Web Server Port  
(not applicable to SPA901)  
Port number of the Linksys IP phone administration web server.  
Defaults to 80.  
Enable Web Admin Access  
(not applicable to SPA901)  
Lets you enable or disable local access to the administration web server. Select yes or no  
from the drop-down menu.  
Defaults to yes.  
Admin Passwd  
(not applicable to SPA901)  
Password for the administrator.  
Defaults to no password.  
User Password  
Password for the user.  
Defaults to no password.  
Internet Connection Type  
Field  
Description  
DHCP  
Enable or disable DHCP.  
Defaults to DHCP (yes on the SPA901).  
Static IP  
IP address. Blank of DHCP assigned address is used.  
(SPA901 only)  
Netmask  
Network mask. Blank if DHCP assigned.  
(SPA901 only)  
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System Tab  
Field  
Description  
Default router IP address. Blank if DHCP assigned.  
Gateway  
(SPA901 only)  
Static IP Settings  
Note  
None of the following parameters apply to the SPA901.  
Field  
Description  
Static IP  
Netmask  
Gateway  
IP address. Blank of DHCP assigned address is used.  
Network mask. Blank if DHCP assigned.  
Default router IP address. Blank if DHCP assigned.  
PPPoE Settings  
Note  
None of the following parameters apply to the SPA901.  
Field  
Description  
PPPoE Login Name  
PPPoE Login Password  
PPPoE Service Name  
Specifies the account name assigned by the ISP for connecting on a Point-to-Point  
Protocol over Ethernet (PPPoE) link.  
Specifies the password assigned by the ISP for connecting on a Point-to-Point Protocol  
over Ethernet (PPPoE) link.  
Specifies the service name assigned by the ISP for connecting on a Point-to-Point Protocol  
over Ethernet (PPPoE) link.  
Optional Network Configuration  
Field  
Description  
Host Name  
Domain  
The host name of the Linksys IP phone.  
The network domain of the Linksys IP phone.  
Primary DNS  
DNS server used by Linksys IP phone in addition to DHCP supplied DNS servers if  
DHCP is enabled; when DHCP is disabled, this is the primary DNS server.  
Defaults to 0.0.0.0.  
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System Tab  
Field  
Description  
Secondary DNS  
DNS server used by Linksys IP phone in addition to DHCP supplied DNS servers if  
DHCP is enabled; when DHCP is disabled, this is the secondary DNS server.  
Defaults to 0.0.0.0.  
DNS Server Order  
(Not applicable to SPA901)  
Specifies the method for selecting the DNS server. The options are Manual,  
Manual/DHCP, and DHCP/Manual.  
DNS Query Mode  
Do parallel or sequential DNS Query. With parallel DNS query mode, the Linksys IP  
phone sends the same request to all the DNS servers at the same time when doing a DNS  
lookup, the first incoming reply is accepted by the Linksys IP phone.  
Defaults to parallel.  
Syslog Server  
Debug Server  
Debug Level  
Specify the syslog server name and port. This feature specifies the server for logging  
Linksys IP phone system information and critical events. If both Debug Server and Syslog  
Server are specified, Syslog messages are also logged to the Debug Server.  
The debug server name and port. This feature specifies the server for logging Linksys IP  
phone debug information. The level of detailed output depends on the debug level  
parameter setting.  
The debug level from 0-3. The higher the level, the more debug information is generated.  
Zero (0) means no debug information is generated. To log SIP messages, you must set the  
Debug Level to at least 2.  
Defaults to 0.  
Primary NTP Server  
IP address or name of primary NTP server.  
(Not applicable to SPA901)  
Secondary NTP Server  
IP address or name of secondary NTP server.  
(Not applicable to SPA901)  
VLAN Settings  
Note  
None of the following parameters apply to the SPA901.  
Field  
Description  
Enable VLAN  
Enable this parameter if the switch to which the Linksys IP phone is connected uses  
VLAN tagging.  
VLAN ID  
Enter the VLAN tag for the VLAN to which the Linksys IP phone is assigned. This should  
be the same VLAN to which the SPA9000 is assigned.  
Enable CDP  
Enables or disables Cisco Discovery Protocol (CDP.  
Defaults to Yes.  
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SIP Tab  
SIP Tab  
This section describes the fields for the following headings on the SIP tab:  
SIP Parameters  
Note  
None of the following parameters apply to the SPA901.  
Field  
Description  
Max Forward  
SIP Max Forward value, which can range from 1 to 255.  
Defaults to 70.  
Max Redirection  
Max Auth  
Number of times an invite can be redirected to avoid an infinite loop.  
Defaults to 5.  
Maximum number of times (from 0 to 255) a request may be challenged.  
Defaults to 2.  
SIP User Agent Name  
Used in outbound REGISTER requests.  
Defaults to $VERSION. If empty, the header is not included. Macro expansion of $A to  
$D corresponding to GPP_A to GPP_D allowed.  
SIP Server Name  
Server header used in responses to inbound responses.  
Defaults to $VERSION.  
SIP Reg User Agent Name  
User-Agent name to be used in a REGISTER request. If this is not specified, the <SIP User  
Agent Name> is also used for the REGISTER request.  
Defaults to blank.  
SIP Accept Language  
Accept-Language header used. To access, click the SIP tab, and fill in the SIP Accept  
Language field.  
There is no default (this indicates Linksys IP phone does not include this header). If  
empty, the header is not included.  
DTMF Relay MIME Type  
MIME Type used in a SIP INFO message to signal a DTMF event. This field must match  
that of the Service Provider.  
Defaults to application/dtmf-relay.  
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Chapter 5 Linksys IP Phone Field Reference  
SIP Tab  
Field  
Description  
Remove Last Reg  
Lets you remove the last registration before registering a new one if the value is different.  
Select yes or no from the drop-down menu.  
Defaults to no.  
Use Compact Header  
Lets you use compact SIP headers in outbound SIP messages. Select yes or no from the  
drop-down menu. If set to yes, the Linksys IP phone uses compact SIP headers in  
outbound SIP messages. If set to no, the Linksys IP phone uses normal SIP headers. If  
inbound SIP requests contain compact headers, Linksys IP phone reuses the same compact  
headers when generating the response regardless the settings of the <Use Compact  
Header> parameter. If inbound SIP requests contain normal headers, Linksys IP phone  
substitutes those headers with compact headers (if defined by RFC 261) if <Use Compact  
Header> parameter is set to yes.  
Defaults to no.  
Escape Display Name  
Lets you keep the Display Name private. Select yes if you want the Linksys IP phone to  
enclose the string (configured in the Display Name) in a pair of double quotes for  
outbound SIP messages. Any occurrences of or \ in the string is escaped with \ and \\ inside  
the pair of double quotes. Otherwise, select no.  
Defaults to yes.  
SIP-B Enable  
Talk Package  
Enables Sylantro call features.  
Enables support for the Broadsoft Talk Package, which enables a user to answer or resume  
a call by clicking a button in an external application.  
Hold Package  
Enables support for the Broadsoft Hold Package, which enables a user to place a call on  
hold by clicking a button in an external application.  
Conference Package  
Notify Conference  
RFC 2543 Call Hold  
Enables support for the Broadsoft Conference Package, which enables a user to start a  
conference by clicking a button in an external application.  
If enabled, the unit will send out a NOTIFY with event=conference when starting a  
conference.  
If set to yes, unit will include c=0.0.0.0 syntax in SDP when sending a SIP re-INVITE to  
the peer to hold the call. If set to no, unit will not include the c=0.0.0.0 syntax in the SDP.  
The unit will always include a=sendonly syntax in the SDP in either case.  
Defaults to yes.  
Random REG CID On Reboot  
If set to yes, the Linksys IP phone uses a different random call-ID for registration after the  
next software reboot. If set to no, the Linksys IP phone tries to use the same call-ID for  
registration after the next software reboot. The Linksys IP phone always uses a new  
random Call-ID for registration after a power-cycle, regardless of this setting.  
Defaults to no.  
SIP TCP Port Min  
SIP TCP Port Max  
CTI Enable  
Specifies the lowest TCP port number that can be used for SIP sessions. Defaults to 5060.  
Specifies the highest TCP port number that can be used for SIP sessions. Defaults to 5080.  
Enables or disables the Computer Telephone Interface feature provided by some servers.  
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SIP Tab  
SIP Timer Values (sec)  
Note  
None of the following parameters apply to the SPA901.  
Field  
Description  
SIP T1  
RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds. Defaults to .5  
seconds.  
SIP T2  
SIP T4  
RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE  
responses), which can range from 0 to 64 seconds. Defaults to 4 seconds.  
RFC 3261 T4 value (maximum duration a message remains in the network), which can  
range from 0 to 64 seconds. Defaults to 5 seconds.  
SIP Timer B  
SIP Timer F  
INVITE time-out value, which can range from 0 to 64 seconds. Defaults to 16 seconds.  
Non-INVITE time-out value, which can range from 0 to 64 seconds.Defaults to 16  
seconds.  
SIP Timer H  
INVITE final response, time-out value, which can range from 0 to 64 seconds.Defaults to  
16 seconds.  
SIP Timer D  
SIP Timer J  
ACK hang-around time, which can range from 0 to 64 seconds. Defaults to 16 seconds.  
Non-INVITE response hang-around time, which can range from 0 to 64 seconds. Defaults  
to 16 seconds.  
INVITE Expires  
ReINVITE Expires  
Reg Min Expires  
INVITE request Expires header value. If you enter 0, the Expires header is not included  
in the request. Ranges from 0 to 19999999999999999999999999999999. Defaults to 240.  
ReINVITE request Expires header value. If you enter 0, the Expires header is not included  
in the request. Ranges from 0 to 19999999999999999999999999999999. Defaults to 30.  
Minimum registration expiration time allowed from the proxy in the Expires header or as  
a Contact header parameter. If the proxy returns a value less than this setting, the  
minimum value is used.  
Defaults to 1.  
Reg Max Expires  
Reg Retry Intvl  
Maximum registration expiration time allowed from the proxy in the Min-Expires header.  
If the value is larger than this setting, the maximum value is used.  
Defaults to 7200.  
Interval to wait before the Linksys IP phone retries registration after failing during the last  
registration.  
Defaults to 30.  
Reg Retry Long Intvl  
When registration fails with a SIP response code that does not match<Retry Reg RSC>,  
the Linksys IP phone waits for the specified length of time before retrying. If this interval  
is 0, the Linksys IP phone stops trying. This value should be much larger than the Reg  
Retry Intvl value, which should not be 0.  
Defaults to 1200.  
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Chapter 5 Linksys IP Phone Field Reference  
SIP Tab  
Field  
Reg Retry Random Delay  
Description  
Random delay range (in seconds) to add to <Register Retry Intvl> when retrying  
REGISTER after a failure. This feature was added in Release 5.1.  
Defaults to blank, which disables this feature.  
Reg Retry Long Random Delay Random delay range (in seconds) to add to <Register Retry Long Intvl> when retrying  
REGSITER after a failure. This feature was added in Release 5.1.  
Defaults to blank, which disables this feature.  
Reg Retry Intvl Cap  
The maximum value to cap the exponential back-off retry delay (which starts at <Register  
Retry Intvl> and doubles on every REGISTER retry after a failure). In other words, the  
retry interval is always at <Register Retry Intvl> seconds after a failure. If this feature is  
enabled, <Reg Retry Random Delay> is added on top of the exponential back-off adjusted  
delay value. This feature was added in Release 5.1.  
Defaults to blank, which disables the exponential back-off feature.  
Sub Min Expires  
Sub Max Expires  
Sub Retry Intvl  
This value sets the lower limit of the REGISTER expires value returned from the Proxy  
server.  
This value sets the upper limit of the REGISTER min-expires value returned from the  
Proxy server in the Min-Expires header. Defaults to 7200.  
This value (in seconds) determines the retry interval when the last Subscribe request fails.  
Defaults to 10.  
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SIP Tab  
Response Status Code Handling  
Field  
Description  
SIT1 RSC  
SIP response status code for the appropriate Special Information Tone (SIT). For example,  
if you set the SIT1 RSC to 404, when the user makes a call and a failure code of 404 is  
returned, the SIT1 tone is played. Reorder or Busy Tone is played by default for all  
unsuccessful response status code for SIT 1 RSC through SIT 4 RSC.  
Defaults to blank.  
SIT2 RSC  
SIT3 RSC  
SIT4 RSC  
SIP response status code to INVITE on which to play the SIT2 Tone.  
Defaults to blank.  
SIP response status code to INVITE on which to play the SIT3 Tone.  
Defaults to blank.  
SIP response status code to INVITE on which to play the SIT4 Tone.  
Defaults to blank.  
Try Backup RSC  
SIP response code that retries a backup server for the current request.  
Defaults to blank.  
(not applicable to SPA901)  
Retry Reg RSC  
Interval to wait before the Linksys IP phone retries registration after failing during the last  
registration.  
Defaults to blank.  
(not applicable to SPA901)  
RTP Parameters  
Field  
Description  
RTP Port Min  
Minimum port number for RTP transmission and reception. <RTP Port Min> and <RTP  
Port Max> should define a range that contains at least 4 even number ports, such as 100 – 106.  
Defaults to 16384.  
RTP Port Max  
Maximum port number for RTP transmission and reception.  
Defaults to 16482.  
RTP Packet SizeSPA901  
(not applicable to SPA901)  
Packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple  
of 0.01 seconds.  
Defaults to 0.030.  
Max RTP ICMP Err  
Number of successive ICMP errors allowed when transmitting RTP packets to the peer  
before the Linksys IP phone terminates the call. If value is set to 0, the Linksys IP phone  
ignores the limit on ICMP errors.  
(not applicable to SPA901)  
Defaults to 0.  
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Chapter 5 Linksys IP Phone Field Reference  
SIP Tab  
Field  
Description  
RTCP Tx Interval  
Interval for sending out RTCP sender reports on an active connection. It can range from  
0 to 255 seconds. During an active connection, the Linksys IP phone can be programmed  
to send out compound RTCP packet on the connection. Each compound RTP packet  
except the last one contains a SR (Sender Report) and a SDES (Source Description). The  
last RTCP packet contains an additional BYE packet. Each SR except the last one contains  
exactly 1 RR (Receiver Report); the last SR carries no RR. The SDES contains CNAME,  
NAME, and TOOL identifiers. The CNAME is set to <User ID>@<Proxy>, NAME is set  
to <Display Name> (or Anonymous if user blocks caller ID), and TOOL is set to the  
Vendor/Hardware-platform-software-version (such as Linksys/Linksys IP  
(not applicable to SPA901)  
phone-1.0.31(b)). The NTP timestamp used in the SR is a snapshot of the Linksys IP  
phone’s local time, not the time reported by an NTP server. If the Linksys IP phone  
receives a RR from the peer, it attempts to compute the round trip delay and show it as the  
<Call Round Trip Delay> value (ms) in the Info section of Linksys IP phone web page.  
Defaults to 0.  
No UDP Checksum  
Select yes if you want the Linksys IP phone to calculate the UDP header checksum for  
SIP messages. Otherwise, select no.  
(not applicable to SPA901)  
Defaults to no.  
Symmetric RTP  
Enable symmetric RTP operation. If enabled, sends RTP packets to the source address and  
port of the last received valid inbound RTP packet. If disabled (or before the first RTP  
packet arrives) sends RTP to the destination as indicated in the inbound SDP.  
(not applicable to SPA901)  
Defaults to yes.  
Stats In BYE  
Determines whether the Linksys IP phone includes the P-RTP-Stat header or response to  
a BYE message. The header contains the RTP statistics of the current call. Select yes or  
no from the drop-down menu. The format of the P-RTP-Stat header is:  
(not applicable to SPA901)  
P-RTP-State: PS=<packets sent>,OS=<octets sent>,PR=<packets received>,OR=<octets  
received>,PL=<packets lost>,JI=<jitter in ms>,LA=<delay in ms>,DU=<call duration in  
s>,EN=<encoder>,DE=<decoder>.  
Defaults to no.  
SDP Payload Types  
The configured dynamic payloads are used for outbound calls only where the Linksys IP phone presents  
the SDP offer. For inbound calls with a SDP offer, the Linksys IP phone follow the caller dynamic  
payload type assignments.  
The Linksys IP phone uses the configured codec names in its outbound SDP. The Linksys IP phone  
ignores the codec names in incoming SDP for standard payload types (0 – 95). For dynamic payload  
types, the Linksys IP phone identifies the codec by the configured codec names. Comparison is  
case-insensitive.  
Field  
AVT Dynamic Payload  
Description  
AVT dynamic payload type. Ranges from 96-127.  
Defaults to 101.  
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Chapter 5 Linksys IP Phone Field Reference  
SIP Tab  
Field  
Description  
INFOREQ Dynamic Payload  
(not applicable to SPA901)  
INFOREQ dynamic payload type.  
Defaults to blank.  
G726r16 Dynamic Payload  
G.726-16 dynamic payload type. Ranges from 96-127.  
Defaults to 98.  
G726r24 Dynamic Payload  
G.726-24 dynamic payload type. Ranges from 96-127.  
Defaults to 97.  
G726r32 Dynamic Payload  
(not applicable to SPA901)  
G726r32 dynamic payload type.  
Defaults to 2.  
G726r40 Dynamic Payload  
G.726-40 dynamic payload type. Ranges from 96-127.  
Defaults to 96.  
G729b Dynamic Payload  
G.729b dynamic payload type. Ranges from 96-127.  
Defaults to 99.  
EncapRTP Dynamic Payload  
(not applicable to SPA901)  
EncapRTP Dynamic Payload type.  
Defaults to 112.  
RTP-Start-Loopback Dynamic  
(not applicable to SPA901)  
RTP-Start-Loopback Dynamic Payload.  
Defaults to 113.  
RTP-Start-Loopback Codec  
(not applicable to SPA901)  
RTP-Start-Loopback Codec. Select one of following: G711u, G711a, G726-16, G726-24,  
G726-32, G726-40, G729a, or G723.  
Defaults to G711u.  
AVT Codec Name  
AVT codec name used in SDP.  
Defaults to telephone-event.  
(not applicable to SPA901)  
G711u Codec Name  
G.711u codec name used in SDP.  
Defaults to PCMU.  
(not applicable to SPA901)  
G711a Codec Name  
G.711a codec name used in SDP.  
Defaults to PCMA.  
(not applicable to SPA901)  
G726r16 Codec Name  
G.726-16 codec name used in SDP.  
Defaults to G726-16.  
(not applicable to SPA901)  
G726r24 Codec Name  
G.726-24 codec name used in SDP.  
Defaults to G726-24.  
(not applicable to SPA901)  
G726r32 Codec Name  
G.726-32 codec name used in SDP.  
Defaults to G726-32.  
(not applicable to SPA901)  
G726r40 Codec Name  
G.726-40 codec name used in SDP.  
Defaults to G726-40.  
(not applicable to SPA901)  
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SIP Tab  
Field  
Description  
G729a Codec Name  
G.729a codec name used in SDP.  
Defaults to G729a.  
(not applicable to SPA901)  
G729b Codec Name  
G.729b codec name used in SDP.  
Defaults to G729ab.  
(not applicable to SPA901)  
G723 Codec Name  
G.723 codec name used in SDP.  
Defaults to G723.  
(not applicable to SPA901)  
EncapRTP Codec Name  
EncapRTP codec name used in SDP.  
Defaults to encaprtp.  
(not applicable to SPA901)  
NAT Support Parameters  
Note  
None of the following parameters apply to the SPA901.  
Field  
Description  
Handle VIA received  
If you select yes, the Linksys IP phone processes the received parameter in the VIA header  
(this is inserted by the server in a response to any of its requests). If you select no, the  
parameter is ignored. Select yes or no from the drop-down menu.  
Defaults to no.  
Handle VIA rport  
If you select yes, the Linksys IP phone processes the rport parameter in the VIA header  
(this is inserted by the server in a response to any of its requests). If you select no, the  
parameter is ignored. Select yes or no from the drop-down menu.  
Defaults to no.  
Insert VIA received  
Insert VIA rport  
Inserts the received parameter into the VIA header of SIP responses if the received-from  
IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu.  
Defaults to no.  
Inserts the rport parameter into the VIA header of SIP responses if the received-from IP  
and VIA sent-by IP values differ. Select yes or no from the drop-down menu.  
Defaults to no.  
Substitute VIA Addr  
Send Resp To Src Port  
Lets you use NAT-mapped IP:port values in the VIA header. Select yes or no from the  
drop-down menu.  
Defaults to no.  
Sends responses to the request source port instead of the VIA sent-by port. Select yes or  
no from the drop-down menu.  
Defaults to no.  
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Chapter 5 Linksys IP Phone Field Reference  
SIP Tab  
Field  
Description  
STUN Enable  
Enables the use of STUN to discover NAT mapping. Select yes or no from the drop-down  
menu.  
Defaults to no.  
STUN Test Enable  
If the STUN Enable feature is enabled and a valid STUN server is available, the Linksys  
IP phone can perform a NAT-type discovery operation when it powers on. It contacts the  
configured STUN server, and the result of the discovery is reported in a Warning header  
in all subsequent REGISTER requests. If the Linksys IP phone detects symmetric NAT or  
a symmetric firewall, NAT mapping is disabled.  
Defaults to no.  
STUN Server  
EXT IP  
IP address or fully-qualified domain name of the STUN server to contact for NAT  
mapping discovery.  
External IP address to substitute for the actual IP address of the Linksys IP phone in all  
outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.  
If this parameter is specified, the Linksys IP phone assumes this IP address when  
generating SIP messages and SDP (if NAT Mapping is enabled for that line). However,  
the results of STUN and VIA received parameter processing, if available, supersede this  
statically configured value.  
Defaults to blank.  
EXT RTP Port Min  
External port mapping number of the RTP Port Min. number. If this value is not zero, the  
RTP port number in all outgoing SIP messages is substituted for the corresponding port  
value in the external RTP port range.  
Defaults to blank.  
NAT Keep Alive Intvl  
Interval between NAT-mapping keep alive messages.  
Defaults to 15.  
Linksys Key System Parameters  
Field  
Description  
Linksys Key System  
Enable or disable the Linksys Key System on the IP phone.  
Defaults to yes.  
Multicast Address  
Force LAN Codec  
The multicast address is used by the SPA9000 to communicate with the Linksys IP IP  
phones.  
Defaults to 224.168.168.168:6061.  
The choices are: none, G.711u, or G.711a. Defaults to none.  
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Provisioning Tab  
Provisioning Tab  
For information about the Provisioning page, see the Linksys SPA Provisioning Guide.  
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Chapter 5 Linksys IP Phone Field Reference  
Regional Tab  
Regional Tab  
This section describes the fields for the following headings on the Regional tab:  
Call Progress Tones  
Note  
None of the following parameters apply to the SPA901.  
Field  
Description  
Dial Tone  
Prompts the user to enter a phone number.  
Defaults to 350@-19,440@-19;10(*/0/1+2).  
Outside Dial Tone  
Alternative to the Dial Tone. It prompts the user to enter an external phone number, as  
opposed to an internal extension. It is triggered by a, (comma) character encountered in  
the dial plan.  
Defaults to 420@-16;10(*/0/1).  
Prompt Tone  
Busy Tone  
Prompts the user to enter a call forwarding phone number.  
Defaults to 520@-19,620@-19;10(*/0/1+2).  
Played when a 486 RSC is received for an outbound call.  
Defaults to 480@-19,620@-19;10(.5/.5/1+2).  
Reorder Tone  
Played when an outbound call has failed or after the far end hangs up during an established  
call. Reorder Tone is played automatically when <Dial Tone> or any of its alternatives  
times out.  
Defaults to 480@-19,620@-19;10(.25/.25/1+2).  
Off Hook Warning Tone  
Played when the caller has not properly placed the handset on the cradle. Off Hook  
Warning Tone is played when Reorder Tone times out.  
Defaults to 480@10,620@0;10(.125/.125/1+2).  
Ring Back Tone  
Played during an outbound call when the far end is ringing.  
Defaults to 440@-19,480@-19;*(2/4/1+2).  
Call Waiting Tone  
Played when a call is waiting. Defaults to 440@-10;30(.3/9.7/1)  
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Regional Tab  
Field  
Description  
Confirm Tone  
Brief tone to notify the user that the last input value has been accepted.  
Defaults to 600@-16; 1(.25/.25/1).  
SIT1 Tone  
Alternative to the Reorder Tone played when an error occurs as a caller makes an  
outbound call. The RSC to trigger this tone is configurable on the SIP screen.  
Defaults to 985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0).  
SIT2 Tone  
SIT3 Tone  
SIT4 Tone  
Alternative to the Reorder Tone played when an error occurs as a caller makes an  
outbound call. The RSC to trigger this tone is configurable on the SIP screen.  
Defaults to 914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0).  
Alternative to the Reorder Tone played when an error occurs as a caller makes an  
outbound call. The RSC to trigger this tone is configurable on the SIP screen.  
Defaults to 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)  
This is an alternative to the Reorder Tone played when an error occurs as a caller makes  
an outbound call. The RSC to trigger this tone is configurable on the SIP screen.  
Defaults to 985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0).  
MWI Dial Tone  
Played instead of the Dial Tone when there are unheard messages in the caller’s mailbox.  
Defaults to 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2).  
Cfwd Dial Tone  
Played when all calls are forwarded.  
Defaults to 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2).  
Holding Tone  
Informs the local caller that the far end has placed the call on hold.  
Defaults to 600@-19*(.1/.1/1,.1/.1/1,.1/9.5/1).  
Conference Tone  
Secure Call Indication Tone  
Played to all parties when a three-way conference call is in progress.  
Defaults to 350@-19;20(.1/.1/1,.1/9.7/1).  
Played when a call has been successfully switched to secure mode. It should be played  
only for a short while (less than 30 seconds) and at a reduced level (less than -19 dBm) so  
it does not interfere with the conversation.  
Defaults to 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2).  
Page Tone  
Alert Tone  
Specifies the tone transmitted when the paging feature is enabled.  
Defaults to 600@-16;.3(.05/0.05/1).  
Played when an alert occurs.  
Defaults to 600@-19;.2(.05/0.05/1).  
Distinctive Ring Patterns  
Note  
None of the following parameters apply to the SPA901.  
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Regional Tab  
Field  
Description  
Cadence 1  
Cadence script for distinctive ring 1.  
Defaults to 60(2/4).  
Cadence 2  
Cadence 3  
Cadence 4  
Cadence 5  
Cadence 6  
Cadence 7  
Cadence 8  
Cadence 9  
Cadence script for distinctive ring 2.  
Defaults to 60(.3/.2, 1/.2,.3/4.  
Cadence script for distinctive ring 3.  
Defaults to 60(.8/.4,.8/4).  
Cadence script for distinctive ring 4.  
Defaults to 60(.4/.2,.3/.2,.8/4).  
Cadence script for distinctive ring 5.  
Defaults to 60(.2/.2,.2/.2,.2/.2,1/4)  
Cadence script for distinctive ring 6.  
Defaults to60(.2/.4,.2/.4,.2/4).  
Cadence script for distinctive ring 7.  
Defaults to 60(4.5/4).  
Cadence script for distinctive ring 8.  
Defaults to 60(0.25/9.75)  
Cadence script for distinctive ring 9.  
Defaults to 60(.4/.2,.4/2).  
Control Timer Values (sec)  
Field  
Description  
Reorder Delay  
Delay after far end hangs up before reorder tone is played. 0 = plays immediately, inf =  
never plays. Range: 0–255 seconds.  
(Not applicable to SPA901)  
Defaults to 255.  
Call Back Expires  
Expiration time in seconds of a call back activation. Range: 0–65535 seconds.  
Defaults to 1800.  
Call Back Retry Intvl  
Call back retry interval in seconds. Range: 0–255 seconds.  
Defaults to 30.  
Call Back Delay  
Delay after receiving the first SIP 18x response before declaring the remote end is  
ringing. If a busy response is received during this time, the Linksys IP phone still  
considers the call as failed and keeps on retrying.  
(Not applicable to SPA901)  
Defaults to 0.5.  
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Chapter 5 Linksys IP Phone Field Reference  
Regional Tab  
Field  
Description  
Interdigit Long Timer  
Long timeout between entering digits when dialing. The interdigit timer values are used  
as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all  
valid matching sequences in the dial plan are incomplete as dialed. Range: 0–64 seconds.  
(Not applicable to SPA901)  
Defaults to 10.  
Interdigit Short Timer  
Short timeout between entering digits when dialing. The Interdigit_Short_Timer is used  
after any one digit, if at least one matching sequence is complete as dialed, but more  
dialed digits would match other as yet incomplete sequences. Range: 0–64 seconds.  
(Not applicable to SPA901)  
Defaults to 3.  
Vertical Service Activation Codes  
The following Vertical Service Activation Codes are automatically appended to the dial-plan.  
Note  
Some of the following parameters do not apply to the SPA901.  
Field  
Description  
Call Return Code  
This code calls the last caller.  
Defaults to *69.  
Blind Transfer Code  
Begins a blind transfer of the current call to the extension specified after the activation  
code.  
Defaults to *98.  
Call Back Act Code  
Call Back Deact Code  
Cfwd All Act Code  
Cfwd All Deact Code  
Cfwd Busy Act Code  
Starts a callback when the last outbound call is not busy.  
Defaults to *66.  
Cancels a callback.  
Defaults to *86.  
Forwards all calls to the extension specified after the activation code.  
Defaults to *72.  
Cancels call forwarding of all calls.  
Defaults to *73.  
Forwards busy calls to the extension specified after the activation code.  
Defaults to *90.  
Cfwd Busy Deact Code  
Cancels call forwarding of busy calls.  
Defaults to *91.  
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Chapter 5 Linksys IP Phone Field Reference  
Regional Tab  
Field  
Description  
Forwards no-answer calls to the extension specified after the activation code.  
Cfwd No Ans Act Code  
Defaults to *92.  
Cfwd No Ans Deact Code  
CW Act Code  
Cancels call forwarding of no-answer calls.  
Defaults to *93.  
Enables call waiting on all calls.  
Defaults to *56.  
CW Deact Code  
Disables call waiting on all calls.  
Defaults to *57.  
CW Per Call Act Code  
CW Per Call Deact Code  
Block CID Act Code  
Block CID Deact Code  
Block CID Per Call Act Code  
Enables call waiting for the next call.  
Defaults to *71.  
Disables call waiting for the next call.  
Defaults to *70.  
Blocks caller ID on all outbound calls.  
Defaults to *67.  
Removes caller ID blocking on all outbound calls.  
Defaults to *68.  
Blocks caller ID on the next outbound call.  
Defaults to *81.  
Block CID Per Call Deact Code Removes caller ID blocking on the next inbound call.  
Defaults to *82.  
Block ANC Act Code  
Block ANC Deact Code  
DND Act Code  
Blocks all anonymous calls.  
Defaults to *77.  
Removes blocking of all anonymous calls.  
Defaults to *87.  
Enables the do not disturb feature.  
Defaults to *78.  
DND Deact Code  
Disables the do not disturb feature.  
Defaults to *79.  
Secure All Call Act Code  
Secure No Call Act Code  
Makes all outbound calls secure.  
Defaults to *16.  
Makes all outbound calls not secure.  
Defaults to *17.  
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Chapter 5 Linksys IP Phone Field Reference  
Regional Tab  
Field  
Description  
Secure One Call Act Code  
Makes the next outbound call secure. (It is redundant if all outbound calls are secure by  
default.)  
Defaults to *18.  
Secure One Call Deact Code  
Makes the next outbound call not secure. (It is redundant if all outbound calls are not  
secure by default.)  
Defaults to *19.  
Paging Code  
The star code used for paging the other clients in the group.  
Defaults to *96.  
Call Park Code  
The star code used for parking the current call.  
Defaults to *38.  
Call Pickup Code  
Call UnPark Code  
Group Call Pickup Code  
Media Loopback Code  
Referral Services Codes  
The star code used for picking up a ringing call.  
Defaults to *36.  
The star code used for picking up a call from the call park.  
Defaults to *39.  
The star code used for picking up a group call.  
Defaults to *37.  
The star code used for media loopback.  
Defaults to *03.  
These codes tell the Linksys IP phone what to do when the user places the current call on  
hold and is listening to the second dial tone.  
One or more *code can be configured into this parameter, such as *98, or *97|*98|*123,  
etc. Max total length is 79 chars. This parameter applies when the user places the current  
call on hold (by Hook Flash) and is listening to second dial tone. Each *code (and the  
following valid target number according to current dial plan) entered on the second  
dial-tone triggers the Linksys IP phone to perform a blind transfer to a target number that  
is prepended by the service *code.  
For example, after the user dials *98, the Linksys IP phone plays a special dial tone called  
the Prompt Tone while waiting for the user the enter a target number (which is checked  
according to dial plan as in normal dialing). When a complete number is entered, the  
Linksys IP phone sends a blind REFER to the holding party with the Refer-To target  
equals to *98<target_number>. This feature allows the Linksys IP phone to hand off a call  
to an application server to perform further processing, such as call park.  
The *codes should not conflict with any of the other vertical service codes internally  
processed by the Linksys IP phone. You can empty the corresponding *code that you do  
not want to Linksys IP phone to process.  
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Chapter 5 Linksys IP Phone Field Reference  
Regional Tab  
Field  
Description  
Feature Dial Services Codes  
These codes tell the Linksys IP phone what to do when the user is listening to the first or  
second dial tone.  
One or more *code can be configured into this parameter, such as *72, or  
*72|*74|*67|*82, etc. Max total length is 79 chars. This parameter applies when the user  
has a dial tone (first or second dial tone). Enter *code (and the following target number  
according to current dial plan) entered at the dial tone triggers the Linksys IP phone to  
call the target number prepended by the *code. For example, after user dials *72, the  
Linksys IP phone plays a prompt tone awaiting the user to enter a valid target number.  
When a complete number is entered, the Linksys IP phone sends a INVITE to  
*72<target_number> as in a normal call. This feature allows the proxy to process features  
like call forward (*72) or BLock Caller ID (*67).  
The *codes should not conflict with any of the other vertical service codes internally  
processed by the Linksys IP phone. You can empty the corresponding *code that you do  
not want to Linksys IP phone to process.  
You can add a parameter to each *code in Features Dial Services Codes to indicate what  
tone to play after the *code is entered, such as *72‘c‘|*67‘p‘. Below are a list of allowed  
tone parameters (note the use of back quotes surrounding the parameter w/o spaces)  
‘c‘ = <Cfwd Dial Tone>  
‘d‘ = <Dial Tone>  
‘m‘ = <MWI Dial Tone>  
‘o‘ = <Outside Dial Tone>  
‘p‘ = <Prompt Dial Tone>  
‘s‘ = <Second Dial Tone>  
‘x‘ = No tones are place, x is any digit not used above  
If no tone parameter is specified, the Linksys IP phone plays Prompt tone by default.  
If the *code is not to be followed by a phone number, such as *73 to cancel call  
forwarding, do not include it in this parameter. In that case, simple add that *code in the  
dial plan and the Linksys IP phone send INVITE *73@..... as usual when user dials *73.  
Vertical Service Announcement Codes  
Note  
None of the following parameters apply to the SPA901.  
Service Annc (Annoucement) Base Number: Defaults to blank.  
Service Annc (Announcement) Extension Codes: Defaults to blank.  
Outbound Call Codec Selection Codes  
These codes automatically appended to the dial-plan. So no need to include them in dial-plan (although  
no harm to do so either).  
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Chapter 5 Linksys IP Phone Field Reference  
Regional Tab  
Note  
None of the following parameters apply to the SPA901.  
Field  
Description  
Prefer G711u Code  
Force G711u Code  
Prefer G711a Code  
Force G711a Code  
Prefer G723 Code  
Makes this codec the preferred codec for the associated call.  
Defaults to *017110.  
Makes this codec the only codec that can be used for the associated call.  
Defaults to *027110.  
Makes this codec the preferred codec for the associated call.  
Defaults to *017111  
Makes this codec the only codec that can be used for the associated call.  
Defaults to *027111.  
Makes this codec the preferred codec for the associated call.  
Defaults to *01723.  
Force G723 Code  
Makes this codec the only codec that can be used for the associated call.  
Defaults to *02723.  
Prefer G726r16 Code  
Force G726r16 Code  
Prefer G726r24 Code  
Force G726r24 Code  
Prefer G726r32 Code  
Force G726r32 Code  
Prefer G726r40 Code  
Force G726r40 Code  
Prefer G729a Code  
Makes this codec the preferred codec for the associated call.  
Defaults to *0172616.  
Makes this codec the only codec that can be used for the associated call.  
Defaults to *0272616.  
Makes this codec the preferred codec for the associated call.  
Defaults to *0172624.  
Makes this codec the only codec that can be used for the associated call.  
Defaults to *0272624.  
Makes this codec the preferred codec for the associated call.  
Defaults to *0172632.  
Makes this codec the only codec that can be used for the associated call.  
Defaults to *0272632.  
Makes this codec the preferred codec for the associated call.  
Defaults to *0172640.  
Makes this codec the only codec that can be used for the associated call.  
Defaults to *0272640.  
Makes this codec the preferred codec for the associated call.  
Defaults to *01729.  
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Regional Tab  
Field  
Description  
Force G729a Code  
Makes this codec the only codec that can be used for the associated call.  
Defaults to *02729.  
Miscellaneous  
Note  
Only the first four parameters apply to the SPA901:  
Set Local Date  
Set Local Time  
Time Zone  
Time Offset  
Field  
Description  
Set Local Date (mm/dd)  
Set Local Time (HH/mm)  
Time Zone  
Sets the local date (mm represents the month and dd represents the day). The year is  
optional and uses two or four digits.  
Sets the local time (hh represents hours and mm represents minutes). Seconds are  
optional.  
Selects the number of hours to add to GMT to generate the local time for caller ID  
generation. Choices are GMT-12:00, GMT-11:00,…, GMT, GMT+01:00, GMT+02:00,  
…, GMT+13:00.  
Defaults to GMT-08:00.  
Time Offset (HH/mm)  
This specifies the offset from GMT to use for the local system time.  
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Regional Tab  
Field  
Description  
Enter the rule for calculating daylight saving time; it should include the start, end, and save  
values. This rule is comprised of three fields. Each field is separated by ; (a semicolon) as  
shown below. Optional values inside [ ] (the brackets) are assumed to be 0 if they are not  
specified. Midnight is represented by 0:0:0 of the given date.  
Daylight Saving Time Rule  
This is the format of the rule: Start = <start-time>; end=<end-time>; save = <save-time>.  
The <start-time> and <end-time> values specify the start and end dates and times of  
daylight saving time. Each value is in this format: <month> /<day> /  
<weekday>[/HH:[mm[:ss]]]  
The <save-time> value is the number of hours, minutes, and/or seconds to add to the  
current time during daylight saving time. The <save-time> value can be preceded by  
a negative (-) sign if subtraction is desired instead of addition. The <save-time> value  
is in this format: [/[+|-]HH:[mm[:ss]]]  
The <month> value equals any value in the range 1-12 (January-December).  
The <day> value equals [+|-] any value in the range 1-31.  
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).  
The <weekday> value equals any value in the range 1-7 (Monday-Sunday). It can  
also equal 0. If the <weekday> value is 0, this means that the date to start or end  
daylight saving is exactly the date given. In that case, the <day> value must not be  
negative. If the <weekday> value is not 0 and the <day> value is positive, then  
daylight saving starts or ends on the <weekday> value on or after the date given. If  
the <weekday> value is not 0 and the <day> value is negative, then daylight saving  
starts or ends on the <weekday> value on or before the date given.  
The abbreviation HH stands for hours (0-23).  
The abbreviation mm stands for minutes (0-59).  
The abbreviation ss stands for seconds (0-59).  
The default Daylight Saving Time Rule is start=4/1/7;end=10/-1/7;save=1.  
DTMF Playback Level  
DTMF Playback Length  
Inband DTMF Boost  
Local DTMF playback level in dBm, up to one decimal place.  
Defaults to -16.  
Local DTMF playback duration in milliseconds.  
Defaults to .1.  
Controls the amount of amplification applied DTMF signals.  
Choices are OdB, 3dB, 6dB, 9dB, 12dB, 15dB, or 18dB.  
Defaults to 12dB.  
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Regional Tab  
Field  
Description  
Dictionary Server Script  
Defines 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>  
Defaults to 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>;  
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  
Specifies the default language. The 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>  
Defaults to blank and the maximum number of characters is 512. The following is an  
example:  
<Language_Selection ua="na"> Spanish  
</Language_Selection>  
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Chapter 5 Linksys IP Phone Field Reference  
Phone Tab  
Phone Tab  
This section describes the fields for the following headings on the Phone tab:  
General  
Field  
Description  
Station Name  
Voice Mail Number  
A name to identify this station (reserved for future use).  
A phone number or URL to check voicemail.  
Text Logo  
Sets the text logo to display when the phone boots up. Default is (blank) which displays  
Linksys. Up to 2 lines of text can be setup; each line should contain less than 32  
characters. A newline character (\n) must be inserted between the 2 lines and escaped with  
%0a. For example, “Super%0aTelecom”.  
(Not applicable to SPA901)  
The URL locating the bitmap (.BMP) file to display on the LCD background. The URL must  
include the TFTP server name (or IP address), directory, and filename, for example:  
BMP Picture Download URL  
(Not applicable to SPA901)  
tftp://myserver.mydomain.com/images/downloadablebmppicture.bmp  
For the SPA962, you can use a 320 x 240 pixel .BMP image (1, 4, 8, 24, or 32-bit color) for  
the background picture. For the SPA942, the image should be 128 x 48 black and white (1-bit  
color).  
Select Logo  
Select from Default, BMP Picture, Text Logo, or None.  
Defaults to Default.  
Select Background Picture  
Select from Default, BMP Picture, or None.  
Defaults to Default.  
Line Key 1/2/3/4/5/6  
When used in the configuration profile, parameters in this section must be appended with [n], where n  
is 1, 2, 3, or 4).  
This section does not apply to the WIP310 wireless phone.  
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Phone Tab  
Field  
Description  
Extension  
Select an extension to be used for this line key. Choices are:  
One-line phone: 1 or disabled,  
Four-line phone: 1, 2, 3, 4, or disabled.  
Six-line phone: 1, 2, 3, 4, 5, 6, or disabled.  
Defaults to 1.  
Short Name  
A short label shown on the LCD display for Line key 1/2/3/4/5/6.  
(Not applicable to SPA901)  
Share Call Appearance  
Yes indicates that Line Key 1/2/3/4/5/6 is a shared call appearance. Otherwise this call  
appearance is not shared (in other words it is private).  
(Not applicable to SPA901)  
Defaults to no.  
Miscellaneous Line Key Settings  
This section does not apply to the WIP310 wireless phone.  
Field  
Description  
Specifies the shared call appearance line ID mapping. Choose Vertical First or Horizontal  
First. Each LED can hold two calls and the first call on an LED makes it light up. Horizontal  
first means the second call makes the same LED flash. Vertical first means the second call  
lights up the next LED.  
SCA Line ID Mapping  
For example, if Extension 101 is assigned to two LEDs, and Vertical First is selected, the sec-  
ond call on Extension 101 lights up the second LED. The third call makes the first LED flash,  
and the fourth call makes the second LED flash.  
If Horizontal First is selected, the second call on Extension 101 makes the first LED flash.  
The third call lights up the second LED, and the fourth call makes the second LED flash.  
Enables the SCA Barge-In.  
Defaults to no.  
SCA Barge-In Enable  
Line Key LED Pattern  
Line key LED colors are described in “Line Key LEDs” section on page 3-22.  
Note  
None of the following parameters apply to the SPA901.  
This section does not apply to the WIP310 wireless phone.  
Field  
Description  
Idle LED  
LED pattern during the Idle state, where the call appearance is not is in use and is  
available to make a new call. Leaving this entry blank indicates the default value of c=g.  
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Phone Tab  
Field  
Description  
Remote Undefined LED  
LED pattern during the Remote Undefined state, where the shared call state is undefined  
(the station is still waiting for the state information from the application server). Not  
applicable if the call appearance is not shared. Leaving this entry blank indicates the  
default value of c=r;p=d.  
Local Seized LED  
LED pattern during the Local Seized state, where this station has seized the call  
appearance to prepare for a new outbound call. Leaving this entry blank indicates the  
default value of c=r.  
Remote Seized LED  
Local Progressing LED  
Remote Progressing LED  
LED pattern during the Remote Seized state, where the shared call appearance is seized  
by another station. Not application if the call appearance is not shared. Leaving this entry  
blank indicates the default value of c=r;p=d.  
LED pattern during the Local Progressing state, where this station is attempting on this  
call appearance an outgoing call that is in proceeding (i.e. the called number is ringing).  
Leaving this entry blank indicates the default value of c=r.  
LED pattern during the Remote Progressing state, where another station is attempting on  
this shared call appearance an outbound call that is progressing. Not applicable if the call  
appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d.  
Local Ringing LED  
LED pattern during the Local Ringing state, when the call appearance is ringing. Leaving  
this entry blank indicates the default value of c=r;p=f.  
Remote Ringing LED  
LED pattern during the Remote Ringing state, where the shared call appearance is in  
ringing on another station. Not applicable if the call appearance is not shared. Leaving  
this entry blank indicates the default value of c=r;p=d.  
Local Active LED  
LED pattern during the Local Active state, where the call appearance is engaged in an  
active call. Leaving this entry blank indicates the default value of c=r.  
Remote Active LED  
LED pattern during the Remote Active state, where another station is engaged in an active  
call on this shared call appearance. Not applicable is this call appearance is not shared.  
Leaving this entry blank indicates the default value of c=r;p=d.  
Local Held LED  
LED pattern during the Local Held state, where the call appearance is held by this station.  
Leaving this entry blank indicates the default value of c=r;p=s.  
Remote Held LED  
LED pattern during the Remote Held state, where another station has placed this call  
appearance on hold. Not applicable if the call appearance is not shared. Leaving this entry  
blank indicates the default value of c=4,p=s.  
Register Failed LED  
Disabled LED  
LED pattern when the corresponding extension has failed to register with the proxy  
server. Leaving this entry blank indicates the default value of c=a.  
LED pattern when the Call Appearance is disabled (not available for any incoming or  
outgoing call). Leaving this entry blank indicates the default value of c=o.  
Registering LED  
Call Back Active LED  
LED Pattern when the corresponding extension is trying to register with the proxy server.  
Leaving this entry blank indicates the default value of c=r;p=s.  
Call Back operation is currently active on this call appearance is not shared. Leaving this  
entry blank indicates the default value of c=r;p=s.  
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Phone Tab  
Supplementary Services  
Enable or disable the corresponding supplementary services on the phone. A value of “yes” indicates  
enabled; “no” indicates disabled.  
Note  
Some of the parameters in this section do not apply to the WIP310 wireless phone. None of the following  
parameters apply to the SPA901.  
Field  
Description  
Conference Serv  
Attn Transfer Serv  
Blind Transfer Serv  
DND Serv  
Enable/disable Three way conference service.  
Defaults to yes.  
Enable/disable attended-call-transfer service.  
Defaults to yes.  
Enable/disable blind-call-transfer service.  
Defaults to yes.  
Enable/disable do-not-disturb service.  
Defaults to yes.  
Block ANC Serv  
Call Back Serv  
Block CID Serv  
Secure Call Serv  
Cfwd All Serv  
Enable/disable block-anonymous-call service.  
Defaults to yes.  
Enable/disable call-back (a.k.a. repeating dialing) service.  
Defaults to yes.  
Enable/disable blocking outbound Caller-ID service.  
Defaults to yes.  
Enable/disable secure-call service.  
Defaults to yes.  
Enable/disable call-forward-all service.  
Defaults to yes.  
Cfwd Busy Serv  
Enable/disable call-forward-on-busy service.  
Defaults to yes.  
Cfwd On No Ans Serv  
Paging Serv  
Enable/disable call-forward-on-no-answer service.  
Defaults to yes.  
Enable/disable the paging service.  
Defaults to yes.  
Call Park Serv  
Enable/disable the call park service.  
Defaults to yes.  
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Phone Tab  
Field  
Description  
Call Pick Up Serv  
Enable/disable the call pickup service.  
Defaults to yes.  
ACD Login Serv  
Enable/disable the ACD Login Service, used for call centers. Typically enabled with the  
<SIP-B> parameter.  
Defaults to no.  
Group Call Pick Up Serv  
ACD Ext  
Enable/disable the group call pickup service.  
Defaults to yes.  
The extension used for handling ACD calls. Select from 1, 2, 3, 4, 5, or 6.  
Defaults to 1.  
Service Annc Serv  
Web Serv  
Enable/disable sending announcement requests to a customer-supplied announcement server.  
Defaults to no.  
Enable/disable the web server.  
Defaults to yes.  
Ring Tone  
Each entry defines a ring tone to be used on the phone, with an ID between 1 and 10. The ID can be used  
in a DirEntry to indicate which ring tone to use when the corresponding caller calls.  
Field  
Description  
Ring-1  
Ring-2  
Ring-3  
Ring-4  
Ring-5  
Ring-6  
Ring-7  
Ring-8  
Ring-9  
Ring-10  
Ring tone script for Ring 1. Defaults to n=Classic-1;w=3;c=1.  
Ring tone script for Ring 2. Defaults to n=Classic-2;w=3;c=2.  
Ring tone script for Ring 3. Defaults to n=Classic-3;w=3;c=3.  
Ring tone script for Ring 4. Defaults to n=Classic-4;w=3;c=4.  
Ring tone script for Ring 5. Defaults to n=Simple-1;w=2;c=1.  
Ring tone script for Ring 6. Defaults to n=Simple-2;w=2;c=2.  
Ring tone script for Ring 7. Defaults to n=Simple-3;w=2;c=3.  
Ring tone script for Ring 8. Defaults to n=Simple-4;w=2;c=4.  
Ring tone script for Ring 9. Defaults to n=Simple-5;w=2;c=5.  
Ring tone script for Ring 10. Defaults to n=Office;w=4;c=1.  
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Phone Tab  
Auto Input Gain (dB)  
This section does not apply to the WIP310 wireless phone.  
Field  
Description  
Handset Input Gain  
The amount of amplification to apply to the audio input signal for the handset.  
Defaults to zero.  
Headset Input Gain  
The amount of amplification to apply to the audio input signal for the headset.  
Defaults to zero.  
(Not applicable to SPA901)  
Speakerphone Input Gain  
(Not applicable to SPA901)  
The amount of amplification to apply to the audio input signal for the speakerphone.  
Defaults to zero.  
Accoustic Echo Canceller  
Enable  
Enable or disable the accoustic echo canceller.  
Defaults to yes (enable).  
Extension Mobility  
Note  
Some of the parameters in this section do not apply to the WIP310 wireless phone. None of the following  
parameters apply to the SPA901.  
This section does not apply to the WIP310 wireless phone.  
Field  
Description  
Extension Mobility  
Enable or disable extension mobility.  
Defaults to no (disabled).  
EM User Domain  
The user domain for extension mobility.  
Defaults to blank.  
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Ext Tab  
Ext Tab  
The Ext tabs vary by phone:  
SPA901: Ext 1 tab  
SPA942: Ext 1, 2, 3,and 4 tabs  
SPA962: Ext 1, 2, 3, 4, 5, and 6 tabs  
This section describes the fields for the following Ext tab headings:  
In a configuration profile, the Line parameters must be appended with the appropriate numeral to  
indicate the line to which the setting applies. For example:  
[1] to specify line one  
[2] to specify line two  
General  
Field  
Description  
Line Enable  
To enable this line for service, select yes. Otherwise, select no.  
Defaults to yes.  
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Ext Tab  
Share Line Appearance  
Field  
Description  
Indicates whether this extension is to be shared with other stations or private. If the  
Share Ext  
extension is not shared, then a call appearance assigned to this extension is not shared,  
regardless the setting of <Share Call Appearance> for that call appearance. If the  
extension is shared, then whether or not a call appearance assigned to this extension is  
shared follows the setting of <Share Call Appearance> for that call appearance. The  
choices are shared or private.  
Defaults to shared.  
Shared User ID  
The user identified assigned to the shared line appearance.  
Subscription Expires  
The subscription time with the SIP proxy (SPA9000).  
Defaults to 60 seconds.  
NAT Settings  
Field  
Description  
NAT Mapping Enable  
To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes.  
Otherwise, select no.  
Defaults to no.  
NAT Keep Alive Enable  
To send the configured NAT keep alive message periodically, select yes. Otherwise, select  
no.  
Defaults to no.  
Enter the keep alive message that should be sent periodically to maintain the current NAT  
mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a  
REGISTER message without contact is sent.  
NAT Keep Alive Msg  
(Not applicable to SPA901)  
Defaults to $NOTIFY.  
Destination that should receive NAT keep alive messages. If the value is $PROXY, the  
messages are sent to the current or outbound proxy.  
NAT Keep Alive Dest  
(Not applicable to SPA901)  
Defaults to $PROXY.  
Network Settings  
Note  
Some of the parameters in this section do not apply to the WIP310 wireless phone. None of the following  
parameters apply to the SPA901.  
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Ext Tab  
Field  
Description  
SIP TOS/DiffServ Value  
TOS/DiffServ field value in UDP IP packets carrying a SIP message.  
Defaults to 0x68.  
SIP CoS Value  
CoS value for SIP messages.  
Defaults to 3.  
RTP TOS/DiffServ Value  
RTP CoS Value  
ToS/DiffServ field value in UDP IP packets carrying RTP data.  
Defaults to 0xb8.  
CoS value for RTP data.  
Defaults to 6.  
Network Jitter Level  
Determines how jitter buffer size is adjusted by the Linksys IP phone. Jitter buffer size is  
adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10  
milliseconds + current RTP frame size), whichever is larger, for all jitter level settings.  
However, the starting jitter buffer size value is larger for higher jitter levels. This setting  
controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select  
the appropriate setting: low, medium, high, very high, or extremely high.  
Defaults to high.  
Jitter Buffer Adjustment  
Controls how the jitter buffer should be adjusted. Select the appropriate setting: up and  
down, up only, down only, or disable.  
Defaults to up and down.  
SIP Settings  
Note  
Some of the parameters in this section do not apply to the WIP310 wireless phone. Only the SIP Port  
and SIP Debug Option parameters apply to the SPA901.  
Field  
Description  
SIP Transport  
Select from UDP, TCP, or TLS.  
Defaults to UDP.  
SIP Port  
Port number of the SIP message listening and transmission port.  
Defaults to 5060.  
SIP 100REL Enable  
To enable the support of 100REL SIP extension for reliable transmission of provisional  
responses (18x) and use of PRACK requests, select yes. Otherwise, select no.  
Defaults to no.  
EXT SIP Port  
The external SIP port number.  
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Ext Tab  
Field  
Description  
Auth Resync-Reboot  
If this feature is enabled, the Linksys IP phone authenticates the sender when it receives  
the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes.  
Otherwise, select no.  
Defaults to yes.  
SIP Proxy-Require  
The SIP proxy can support a specific extension or behavior when it sees this header from  
the user agent. If this field is configured and the proxy does not support it, it responds  
with the message, unsupported. Enter the appropriate header in the field provided.  
SIP Remote-Party-ID  
To use the Remote-Party-ID header instead of the From header, select yes. Otherwise,  
select no.  
Defaults to yes.  
Referor Bye Delay  
Controls when the Linksys IP phone sends BYE to terminate stale call legs upon  
completion of call transfers. Multiple delay settings (Referor, Refer Target, Referee, and  
Refer-To Target) are configured on this screen. For the Referor Bye Delay, enter the  
appropriate period of time in seconds.  
Defaults to 4.  
Refer Target Bye Delay  
Sticky 183  
For the Refer Target Bye Delay, enter the appropriate period of time in seconds.  
Defaults to 0.  
If this feature is enabled, the IP telephony ignores further 180 SIP responses after  
receiving the first 183 SIP response for an outbound INVITE. To enable this feature,  
select yes. Otherwise, select no.  
Defaults to no.  
Auth INVITE  
When enabled, authorization is required for initial incoming INVITE requests from the  
SIP proxy.  
Referee Bye Delay  
For the Referee Bye Delay, enter the appropriate period of time in seconds.  
Defaults to 0.  
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Ext Tab  
Field  
Description  
SIP Debug Option  
SIP messages are received at or sent from the proxy listen port. This feature controls  
which SIP messages to log. Choices are as follows:  
none—No logging.  
1-line—Logs the start-line only for all messages.  
1-line excl. OPT—Logs the start-line only for all messages except OPTIONS  
requests/responses.  
1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY  
requests/responses.  
1-line excl. REG—Logs the start-line only for all messages except REGISTER  
requests/responses.  
1-line excl. OPT|NTFY|REG—Logs the start-line only for all messages except  
OPTIONS, NOTIFY, and REGISTER  
requests/responses.  
full—Logs all SIP messages in full text.  
full excl. OPT—Logs all SIP messages in full text except OPTIONS  
requests/responses.  
full excl. NTFY—Logs all SIP messages in full text except NOTIFY  
requests/responses.  
full excl. REG—Logs all SIP messages in full text except REGISTER  
requests/responses.  
full excl. OPT|NTFY|REG—Logs all SIP messages in full text except for OPTIONS,  
NOTIFY, and REGISTER requests/responses.  
Defaults to none.  
Refer-To Target Contact  
To contact the refer-to target, select yes. Otherwise, select no.  
Defaults to no.  
Call Feature Settings  
Note  
Many of the following parameters do not apply to the SPA901.  
Field  
Description  
Blind Attn-Xfer Enable  
Enables the Linksys IP phone to perform an attended transfer operation by ending the  
current call leg and performing a blind transfer of the other call leg. If this feature is  
disabled, the Linksys IP phone performs an attended transfer operation by referring the  
other call leg to the current call leg while maintaining both call legs. To use this feature,  
select yes. Otherwise, select no.  
Defaults to no.  
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Ext Tab  
MOH Server  
User ID or URL of the auto-answering streaming audio server. When only a user ID is  
specified, the current or outbound proxy is contacted. Music-on-hold is disabled if the  
MOH Server is not specified.  
Defaults to imusic when used with a SPA9000 IP PBX.  
Message Waiting  
Indicates whether the Message Waiting Indicator on the phone is lit. This parameter is  
toggled by a message from the SIP proxy to indicate if a message is waiting. You can  
manually modify it to clear or set the flag in the Ext 1-6 tab.  
Setting this value to Yes can activate stutter tone and VMWI signal. This parameter is  
stored in long-term memory and survives after reboot or power cycle.  
Defaults to No.  
Auth Page  
Specifies whether to authenticate the invite before auto answering a page.  
Defaults to No.  
Default Ring  
Type of ring heard. This corresponds to the Ring Tone on the Phone tab. Choose from No  
Ring, 1 through 10, User 1, or User 2.  
Defaults to 1.  
Auth Page Realm  
Identifies the Realm part of the Auth that is accepted when the Auth Page parameter is set  
to Yes. This parameter accepts alphanumeric characters.  
Defaults to blank.  
Conference Bridge URL  
Auth Page Password  
This is the URL used to join into a conference call, generally in the form of the word  
“conference” or “user@IPaddress:port”.  
Defaults to blank.  
Identifies the password used when the Auth Page parameter is set to Yes. This parameter  
accepts alphanumeric characters.  
Defaults to blank.  
Mailbox ID  
Identifies the voicemail box number/ID for the phone.  
Defaults to blank.  
Voice Mail Server  
Identifies the SpecVM server for the phone, generally the IP address and port number of  
the VM server.  
State Agent  
Reserved feature.  
CFWD Notify Serv  
Specifies whether to enable a SIP-B feature regarding the sending of a Notify to the phone  
when a call is forwarded elsewhere.  
Defaults to No.  
CFWD Notifier  
Typically, this field is configured with the SIP proxy information.  
Proxy and Registration  
Note  
Many of the following parameters do not apply to the SPA901.  
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Ext Tab  
Field  
Description  
Proxy  
SIP proxy server and port number set by the Service Provider for all outbound requests.  
For example: 192.168.2.100:6060.  
Use Outbound Proxy  
Enable the use of <Outbound Proxy>. If set to no, <Outbound Proxy> and <Use OB Proxy  
in Dialog) is ignored. Defaults to no.  
Outbound Proxy  
SIP Outbound Proxy Server where all outbound requests are sent as the first hop.  
Use OB Proxy In Dialog  
Whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored  
if <Use Outbound Proxy> is no or <Outbound Proxy> is empty.  
Defaults to yes.  
Register  
Enable periodic registration with the <Proxy>. This parameter is ignored if <Proxy> is not  
specified.  
Defaults to yes.  
Make Call Without Reg  
Register Expires  
Allow making outbound calls without successful (dynamic) registration by the unit. If no,  
the dial tone will not play unless registration is successful.  
Defaults to no.  
Allow answering inbound calls without successful (dynamic) registration by the unit. If proxy  
responded to REGISTER with a smaller Expires value, the phone will renew registration based  
on this smaller value instead of the configured value. If registration failed with an Expires too  
brief error response, the phone will retry with the value given in the Min-Expires header in the  
error response.  
Defaults to 60.  
Ans Call Without Reg  
Use DNS SRV  
If enabled, the user does not have to be registered with the proxy to answer calls.  
Defaults to no.  
Whether to use DNS SRV lookup for Proxy and Outbound Proxy.  
Defaults to no.  
DNS SRV Auto Prefix  
Proxy Fallback Intvl  
If enabled, the phone will automatically prepend the Proxy or Outbound Proxy name with  
_sip._udp when performing a DNS SRV lookup on that name.  
Defaults to no.  
This parameter sets the delay (sec) after which the phone will retry from the highest  
priority proxy (or outbound proxy) servers after it has failed over to a lower priority  
server. This parameter is useful only if the primary and backup proxy server list is  
provided to the phone via DNS SRV record lookup on the server name. (Using multiple  
DNS A record per server name does not allow the notion of priority and so all hosts will  
be considered at the same priority and the phone will not attempt to fall back after a fail  
over).  
Defaults to 3600  
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Ext Tab  
Field  
Description  
Proxy Redundancy Method  
Select Normal or Based on SRV port. The phone creates an internal list of proxies returned  
in the DNS SRV records.  
If you select Normal, the list contains proxies ranked by weight and priority.  
If you select Based on SRV, the phone uses normal, then inspects the port number based  
on the first listed proxy port.  
Defaults to Normal.  
Subscriber Information  
Note  
Many of the following parameters do not apply to the SPA901.  
Field  
Description  
Display Name  
User ID  
Display name for caller ID.  
Extension number for this line.  
Password  
Password for this line.  
Defaults to blank.  
Use Auth ID  
To use the authentication ID and password for SIP authentication, select yes. Otherwise,  
select no to use the user ID and password.  
Defaults to no.  
Auth ID  
Authentication ID for SIP authentication.  
Defaults to blank.  
Mini Certificate  
Base64 encoded of Mini-Certificate concatenated with the 1024-bit public key of the CA  
signing the MC of all subscribers in the group.  
Defaults to blank.  
SRTP Private Key  
Base64 encoded of the 512-bit private key per subscriber for establishment of a secure  
call.  
Defaults to blank.  
Audio Configuration  
A codec resource is considered as allocated if it has been included in the SDP codec list of an active call,  
even though it eventually may not be the one chosen for the connection. So, if the G.729a codec is  
enabled and included in the codec list, that resource is tied up until the end of the call whether or not the  
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Ext Tab  
call actually uses G.729a. If the G.729a resource is already allocated and since only one G.729a resource  
is allowed per device, no other low-bit-rate codec may be allocated for subsequent calls; the only choices  
are G711a and G711u. On the other hand, two G.723.1/G.726 resources are available per device.  
Therefore it is important to disable the use of G.729a in order to guarantee the support of two  
simultaneous uses of the G.723/G.726 codecs.  
Field  
Description  
Preferred Codec  
Preferred codec for all calls. (The actual codec used in a call still depends on the outcome of  
the codec negotiation protocol.) Select one of the following: G711u, G711a, G726-16,  
G726-24, G726-32, G726-40, G729a, or G723.  
Defaults to G711u.  
Silence Supp Enable  
Use Pref Codec Only  
To enable silence suppression so that silent audio frames are not transmitted, select yes. Oth-  
erwise, select no.  
Defaults to no.  
To use only the preferred codec for all calls, select yes. (The call fails if the far end does not  
support this codec.) Otherwise, select no.  
Defaults to no.  
Echo Canc Enable  
Enable the use of the echo canceller.  
Defaults to yes.  
Echo Canc Adapt Enable  
Echo Supp Enable  
Enable the echo canceller to adapt.  
Defaults to yes.  
Enable the use of echo suppressor. If <Echo Canc Enable> is “no,” this parameter is  
ignored.  
Defaults to yes.  
G729a Enable  
To enable the use of the G.729a codec at 8 kbps, select yes. Otherwise, select no.  
Defaults to yes.  
G723 Enable  
To enable the use of the G.723a codec at 6.3 kbps, select yes. Otherwise, select no.  
Defaults to yes.  
G726-16 Enable  
G726-24 Enable  
G726-32 Enable  
G726-40 Enable  
FAX CED Detect Enable  
To enable the use of the G.726 codec at 16 kbps, select yes. Otherwise, select no.  
Defaults to yes.  
To enable the use of the G.726 codec at 24 kbps, select yes. Otherwise, select no.  
Defaults to yes.  
To enable the use of the G.726 codec at 32 kbps, select yes. Otherwise, select no.  
Defaults to yes.  
To enable the use of the G.726 codec at 40 kbps, select yes. Otherwise, select no.  
Defaults to yes.  
To enable detection of the fax Caller-Entered Digits (CED) tone, select yes. Otherwise,  
select no.  
Defaults to yes.  
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Ext Tab  
Field  
Description  
FAX CNG Detect Enable  
FAX Passthru Codec  
Defaults to yes.  
Codec to use for FAX passthru (G711u or G711a).  
Defaults to G711u.  
FAX Codec Symmetric  
FAX Passthru Method  
Force unit to use symmetric codec druing FAX passthru.  
Defaults to yes.  
Choose None, NSE, or ReINVITE.  
Defaults to NSE.  
FAX Process NSE  
Defaults to yes.  
FAX Disable ECAN  
Set to yes to automatically disable Echo Canceller when FAX tone is detected.  
Defaults to no.  
DTMF Tx Method  
Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO, Auto,  
InBand+INFO, or AVT+INFO. InBand sends DTMF using the audio path. AVT sends  
DTMF as AVT events. INFO uses the SIP INFO method. Auto uses InBand or AVT based  
on the outcome of codec negotiation.  
Defaults to Auto.  
Hook Flash Tx Method  
Select the method to signal Hook Flash events:  
None: do not signal hook flash events  
AVT: use RFC2833 AVT (event=16)  
INFO: use SIP INFO method with the signal line “signal = hf” in the message body. The  
MIME type for this message body is taken from the <Hook Flash MIME Type> parame-  
ter.  
Defaults to None.  
Symmetric RTP  
Enable symmetric RTP operation. If enabled, the SPA sends RTP packets to the source  
address of the last received valid inbound RTP packet. If disabled, the SPA sends RTP to  
the destination as indicated in the inbound SDP.  
Defaults to yes.  
Release Unused Codec  
This feature allows the release of codecs not used after codec negotiation on the first call,  
so that other codecs can be used for the second line. To use this feature, select yes.  
Otherwise, select no.  
Defaults to yes.  
A codec resource is considered allocated if it has been included in the SDP codec list of an active call,  
even though it eventually might not be chosen for the connection. If the G.729a codec is enabled and  
included in the codec list, that resource is tied up until the end of the call whether or not the call actually  
uses G.729a. If the G729a resource is already allocated (and since only one G.729a resource is allowed  
per SPA), no other low-bit-rate codec can be allocated for subsequent calls. The only choices are G711a  
and G711u.  
Since two G.723.1/G.726 resources are available per SPA, you should disable the use of G.729a to  
guarantee support for two simultaneous G.723/G.726 codecs.  
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Ext Tab  
Dial Plan  
Note  
The following parameters do not apply to the SPA901.  
The default dial plan script for each line is as follows:  
(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxx|xxxxxxxxxxxx.). The syntax for a dial plan  
expression is as follows:  
Table 5-1 Dial Plan Script Syntax  
Dial Plan Entry  
*xx  
Functionality  
Allow arbitrary 2 digit star code  
Allow x11 sequences  
[3469]11  
0
Operator  
00  
International Operator  
[2-9]xxxxxx  
1xxx[2-9]xxxxxx  
xxxxxxxxxxxx.  
US local number  
US 1 + 10-digit long distance number  
Everything else (Int’l long distance, FWD, ...)  
If IP dialing is enabled, one can dial [user-id@]a.b.c.d[:port], where ‘@’, ‘.’, and ‘:’ are dialed by  
entering *, user-id must be numeric (like a phone number) and a, b, c, d must be between 0 and 255, and  
port must be larger than 255. If port is not given, 5060 is used. Port and User-Id are optional. If the  
user-id portion matches a pattern in the dial plan, then it is interpreted as a regular phone number  
according to the dial plan. The INVITE message, however, is still sent to the outbound proxy if it is  
enabled.  
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Ext Tab  
Field  
Description  
Dial plan script for this line.  
Dial Plan  
The dial plan syntax is expanded in the SPA to allow the designation of three parameters  
to be used with a specific gateway:  
uid – the authentication user-id  
pwd – the authentication password  
nat – if this parameter is present, use NAT mapping  
Each parameter is separated by a semi-colon (;).  
Furthermore, it recognizes gw0, gw1, …, gw4 as the locally configured gateways, where  
gw0 represents the local PSTN gateway in the same unit.  
Example 1:  
*1xxxxxxxxxx<:@fwdnat.pulver.com:5082;uid=jsmith;pwd=xyz  
Example 2:  
*1xxxxxxxxxx<:@fwd.pulver.com;nat;uid=jsmith;pwd=xyz  
Example 3:  
[39]11<:@gw0>  
Enable IP Dialing  
Enable or disable IP dialing.  
Defaults to no.  
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User  
User  
This section describes the fields for the following headings on the User tab:  
Call Forward  
See Vertical Service Activation Codes, page 5-22 for information on call forwarding parameters.  
Speed Dial  
Note  
For the SPA901, speed dial is discussed in Call Forward, page 5-48  
This section does not apply to the WIP310 wireless phone.  
Field  
Description  
Speed Dial 2/3/4/5/6/7/8/9  
Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9.  
Defaults to blank.  
Supplementary Services  
The Linksys IP phone provides native support of a large set of enhanced or supplementary services. All  
of these services are optional. The parameters listed in the following table are used to enable or disable  
a specific supplementary service. A supplementary service should be disabled if a) the user has not  
subscribed for it, or b) the Service Provider intends to support similar service using other means than  
relying on the Linksys IP phone.  
Most supplementary service paramters are listed in Supplementary Services, page 5-33.  
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User  
Web Information Service Settings (SPA962)  
Note  
The following parameters apply only to the SPA962.  
.
Field  
Description  
RSS Feed URL [1-5]  
URLs for Local and World news, Finance, Sports, and Politics.  
For more information, including default URLs, see “RSS Newsfeeds (SPA962)” section  
Feed Data Update Interval(m)  
Scrolling Timer(s)  
Interval at which the news feed is updated.  
Defaults to 15 seconds.  
Scrolling timer.  
Defaults to 0.8 seconds.  
Traffic Service Information Settings (SPA962)  
Note  
The following parameters apply only to the SPA962.  
For more information on using the Web Server, see “Configuring the Web Service” section on page 3-14,  
Field  
Description  
State  
Enter your local state.  
Defaults to CA.  
City)  
Enter your local city.  
Defaults to San Jose.  
Street  
Enter your local street.  
Defaults to Cisco.  
Zip Code  
Enter your local zip code.  
Defaults to 95134.  
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User  
Audio Volume  
Note  
Many of the following parameters do not apply to the SPA901.  
Field  
Description  
Ringer Volume  
Speaker Volume  
Handset Volume  
Headset Volume  
Sets the default volume for the ringer.  
Sets the default volume for the speakerphone.  
Sets the default volume for the handset.  
Sets the default volume for the headset.  
Phone GUI Menu Color Settings (SPA962 only)  
Note  
The following parameters apply only to the SPA962.  
Chose from:  
Light Blue  
Light Green  
Pink  
Silver  
Defaults to Light Blue.  
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932 Tab (SPA962 only)  
932 Tab (SPA962 only)  
This tab includes the following sections:  
General  
Field  
Description  
Subscribe Expires  
Specifies how long the subscription remains valid. After the specified period of time,  
elapses, the SPA932 initiates a new subscription.  
Defaults to 1800.  
Subscribe Retry Interval  
Unit 1 Enable  
Specifies the length of time to wait to try again if subscription fails.  
Enables or disables the first SPA932 unit (each SPA962 can have up to two SPA932s  
attached).  
Subscribe Delay  
Length of delay before attempting to subscribe.  
Defaults to 1.  
Test Mode Enable  
Enables or disables test mode. When test mode is enabled, the LEDs are turned on when  
keys are pressed, going from off to green to red, and back to off. In test mode, when all  
the buttons on the sidecar are returned to off, all the keys become amber. The SPA962  
must be rebooted after the test is completed.  
Unit 2 Enable  
Enables or disables the second SPA932 unit (each SPA962 can have up to two SPA932s  
attached).  
Server Type  
Selects the type of server used.  
Call Pickup Code  
The star code used for picking up a ringing call.  
Defaults to *98.  
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932 Tab (SPA962 only)  
Unit 1  
Field  
Description  
Unit 1 Key 1-32  
This field lets you enter a string that defines the extension and other parameters associated  
with each lighted button on the first SPA932 unit. Keywords and values are case-sensitive.  
The configuration script is composed of the following keywords, followed by an equal  
sign (=) and separated by semicolons (;):  
fnc—defines which of the following functions are enabled for the specified key  
(separate more than one function with a plus sign (+):  
blf—busy lamp field function used for monitoring line activity  
sd—speed dial function  
cp—call pickup (if supported by the SIP proxy server)  
sub—Use this keyword to identify the phones to be monitored.  
usr —Use this keyword to identify the specific users or extensions to be monitored.  
nme—(optional) Use this field with the SPA9000 to identify any alias that has been  
assigned to the extension in the Linksys IP phone configuration.  
vid—(optional) Use this field to identify the SPA962 extension to use with the  
monitored list specified by the sub= keyword, when more than one BLF monitored  
list is configured on the SIP proxy server. The possible values are 1 to 6,  
corresponding to each of the six extensions available on the SPA962.  
Note  
You only need to use the vid= keyword in the first entry assigned to each  
SPA962 extension. Subsequent keys will use the same extension.  
The following is the syntax for a Broadsoft server:  
fnc=enabled_functions;sub=blf_list_uri@broadsoft;usr=dn@broadsoft  
Replace the variables (in italics), as follows:  
enabled_functions with cp for call pickup, sd for speed dial, and blf for busy lamp  
field; separate multiple functions in any order with a plus sign (+)  
blf_list_uri with the URI assigned to the BLF monitoring list on the Broadsoft  
server.  
broadsoft with the IP address or hostname of the Broadsoft server.  
dn with the number associated with the user account on the Monitored Users list.  
For example:  
Note  
The nme keyword is not used because the Broadsoft server automatically uses the  
assigned user account name.  
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932 Tab (SPA962 only)  
Unit 2  
Field  
Description  
See the description for Unit 1.  
Unit 2 Key 1-32  
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SPA932 Status  
SPA932 Status  
This page provides two tabs to display the status of up to two SPA932 units that are supported by a single  
SPA962:  
Unit 1—Displays information about the first SPA932 sidecar unit.  
Unit 2—Displays information about the second SPA932 sidecar unit.  
Each tab provides the read-only fields described in the following table:  
Field  
Description  
Unit Enable  
Subscribe Expires  
Displays if the Unit is enabled or disabled.  
Displays when the current subscription expires. After the subscription expires, the SPA932  
automatically requests a new subscription.  
HW Version  
Unit Online  
Displays the version of the hardware.  
Displays whether the unit is powered on and connected or not.  
Subscribe Retry Interval  
SW Version  
Key Name  
Type  
Displays the length of time the SPA932 waits to try again if subscription fails.  
Displays the version of the software currently running on the unit.  
Displays the name assigned to each key (1-32) on the SPA932 sidecar unit.  
Displays the function enabled for each key (1-32) on the SPA932 sidecar unit.  
Displays the extension assigned to each key (1-32) on the SPA932 sidecar unit.  
Displays the subscribe URI configured for each key (1-32) on the SPA932 sidecar unit.  
Line  
Station  
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A P P E N D I X  
A
Acronyms  
A/D  
Analog To Digital Converter  
ANC  
B2BUA  
BLF  
Anonymous Call  
Back to Back User Agent  
Busy Lamp Field  
Bool  
Boolean Values. Specified as “yes” and “no”, or “1” and “0” in the profile  
Certificate Authority  
CA  
CAS  
CPE Alert Signal  
CDR  
CID  
Call Detail Record  
Caller ID  
CIDCW  
CNG  
CPC  
Call Waiting Caller ID  
Comfort Noise Generation  
Calling Party Control  
CPE  
Customer Premises Equipment  
Call Waiting Caller ID  
CWCID  
CWT  
D/A  
Call Waiting Tone  
Digital to Analog Converter  
decibel  
dB  
dBm  
dB with respect to 1 milliwatt  
Dynamic Host Configuration Protocol  
Domain Name Server  
DHCP  
DNS  
DRAM  
DSL  
Dynamic Random Access Memory  
Digital Subscriber Loop  
DSP  
Digital Signal Processor  
DTAS  
DTMF  
FQDN  
FSK  
Data Terminal Alert Signal (same as CAS)  
Dual Tone Multiple Frequency  
Fully Qualified Domain Name  
Frequency Shift Keying  
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Appendix A Acronyms  
FXS  
Foreign eXchange Station  
Gateway  
GW  
ITU  
International Telecommunication Union  
Hypertext Markup Language  
Hypertext Transfer Protocol  
HTTP over SSL  
HTML  
HTTP  
HTTPS  
ICMP  
IGMP  
ILEC  
IP  
Internet Control Message Protocol  
Internet Group Management Protocol  
Incumbent Local Exchange Carrier  
Internet Protocol  
ISP  
Internet Service Provider  
IP Telephony Service Provider  
Interactive Voice Response  
Local Area Network  
ITSP  
IVR  
LAN  
LBR  
LBRC  
MC  
Low Bit Rate  
Low Bit Rate Codec  
Mini-Certificate  
MGCP  
MOH  
MOS  
ms  
Media Gateway Control Protocol  
Music On Hold  
Mean Opinion Score (1-5, the higher the better)  
Millisecond  
MSA  
MWI  
OSI  
Music Source Adaptor  
Message Waiting Indication  
Open Switching Interval  
Printed Circuit Board  
PCB  
PR  
Polarity Reversal  
PS  
Provisioning Server  
PSQM  
PSTN  
NAT  
OOB  
REQT  
RESP  
RSC  
RTP  
Perceptual Speech Quality Measurement (1-5, the lower the better)  
Public Switched Telephone Network  
Network Address Translation  
Out-of-band  
(SIP) Request Message  
(SIP) Response Message  
(SIP) Response Status Code, such as 404, 302, 600  
Real Time Protocol  
RTT  
Round Trip Time  
SAS  
Streaming Audio Server  
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Appendix A Acronyms  
SDP  
SDRAM  
sec  
Session Description Protocol  
Synchronous DRAM  
seconds  
SIP  
Session Initiation Protocol  
Shared line appearance  
Subscriber Line Interface Circuit  
Service Provider  
SLA  
SLIC  
SP  
SPA  
Linksys Phone Adaptor  
Secure Real-time Transport Protocol  
Secure Socket Layer  
SRTP  
SSL  
TFTP  
TCP  
UA  
Trivial File Transfer Protocol  
Transmission Control Protocol  
User Agent  
uC  
Micro-controller  
UDP  
URL  
VM  
User Datagram Protocol  
Uniform Resource Locator  
Voicemail  
VMWI  
VQ  
Visual Message Waiting Indication/Indicator  
Voice Quality  
WAN  
XML  
Wide Area Network  
Extensible Markup Language  
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A P P E N D I X  
B
Glossary  
ACD (Automatic Call Distribution)—A switching system designed to allocate incoming calls to certain  
positions or agents in the order received and to hold calls not ready to be handled (often with a recorded  
announcement).  
Area code—A 3-digit code used in North America to identify a specific geographic telephone location.  
The first digit can be any number between 2 and 9. The second and third digits can be any number.  
Billing increment—The division by which the call is rounded. In the field it is common to see  
full-minute billing on the local invoice while 6-second rounding is the choice of most long-distance  
providers that bill their customers directly.  
Blocked calls—Caused by an insufficient network facility that does not have enough lines to allow calls  
to reach a given destination. May also pertain to a call from an originating number that is blocked by the  
receiving telephone number.  
Bundled service—Offering various services as a complete package.  
Call completion—The point at which a dialed number is answered.  
Call termination—The point at which a call is disconnected.  
CDR (Call Detail Records)—A software program attached to a VoIP/telephone system that records  
information about the telephone number’s activity.  
Carrier’s carrier—Companies that build fiber optic and microwave networks primarily selling to  
resellers and carriers. Their main focus is on the wholesale and not the retail market.  
Casual access—When customers choose not to use their primary carriers to process the long-distance  
call being made. The customer dials the carrier’s 101XXXX number.  
CO (Central Office)—Switching center for the local exchange carrier.  
Centrex—This service is offered by the LEC to the end user. The feature-rich Centrex line offers the  
same features and benefits as a PBX to a customer without the capital investment or maintenance  
charges. The LEC charges a monthly fee to the customer, who must agree to sign a term agreement.  
Circuits—The communication path(s) that carry calls between two points on a network.  
Customer Premise Equipment—The only part of the telecommunications system that the customer  
comes into direct contact with. Example of such pieces of equipment are telephones, key systems, PBXs,  
voice-mail systems, and call accounting systems as well as wiring telephone jacks. The standard for this  
equipment is set by the FCC, and the equipment is supplied by an interconnect company.  
Dedicated access—Customers have direct access to the long-distance provider via a special circuit (T1  
or private lines). The circuit is hardwired from the customer site to the POP and does not pass through  
the LEC switch. The dial tone is provided from the long-distance carrier.  
Linksys IP Phone Administrator Guide  
Firmware Version 5.1  
B-1  
Download from Www.Somanuals.com. All Manuals Search And Download.  
Appendix B Glossary  
Dedicated Access Line (DAL)—Provided by the local exchange carrier. An access line from the  
customer’s telephone equipment directly to the long-distance company’s switch or POP.  
Demarcation point—This is where the LEC ownership and responsibility (wiring, equipment) ends and  
the customer’s responsibilities begin.  
Direct Inward Dialing (DID)—Allows an incoming call to bypass the attendant and ring directly to an  
extension. Available on most PBX systems and a feature of Centrex service.  
Dual Tone Multifrequency (DTMF)—Better known as the push button keypad. DTMF replaces dial  
pulses with electronically produced tones for network signaling.  
Enhanced service—Services that are provided in addition to basic long distance and accessed by way of  
a touchtone phone through a series of menus.  
Exchange code (NXX)—The first three digits of a phone number.  
Flat-rate pricing—The customer is charged one rate (sometimes two rates, one for peak and one for  
off-peak) rather than a mileage-sensitive program rate.  
IXC (Interexchange Carrier)—A long-distance provider that maintains its own switching equipment.  
IVR (Interactive Voice Response)—Provides a mechanism for information to be stored and retrieved  
using voice and a touchtone telephone.  
Local loop—The local telephone company provides the transmission facility from the customer to the  
telephone company’s office, which is engineered to carry voice and/or data.  
North American Numbering Plan (NANP)—How telephone numbers are identified in North America.  
The telephone number can be identified based on their three separate components: (NPA), (NXX), and  
(XXXX).  
PIN (Personal Identification Code)—A customer calling/billing code for prepaid and pay-as-you-go  
calling cards.  
Private Branch Exchange—Advanced phone system commonly used by the medium to larger customer.  
It allows the customer to perform a variety of in-house routing (inside calling). The dial tone that is heard  
when the customer picks up the phone is an internal dial tone.  
SS7 (900 Series phone Signaling Number 7)—Technology used by large carriers to increase the  
reliability and speed of transmission between switches.  
Switch (switching)—Equipment that connects and routes calls and provides other interim functions such  
as least cost routing, IVR, and voicemail. It performs the “traffic cop” function of telecommunications  
via automated management decisions.  
Touchtone (DTMF)—The tone recognized by a push button (touchtone) telephone.  
Unified messaging— Platform that lets users send, receive, and manage all e-mail, voice, and fax  
messages from any telephone, PC, or information device.  
Voicemail—A system that allows storage and retrieval of voice messages through voice-mail boxes.  
Linksys IP Phone Administrator Guide  
Firmware Version 5.1  
B-2  
Download from Www.Somanuals.com. All Manuals Search And Download.  

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