Cisco Systems IP Phone SPA922 User Manual

Advanced, Affordable, Feature-Rich IP  
Phone for Business or a Home Office  
Stylish and functional in design, the SPA922 IP Phone is ideal for a residence or  
business using a hosted IP telephony service, an IP PBX, or a large scale IP Centrex  
deployment. The SPA922 leverages industry-leading VoIP technology from Linksys to  
deliver an upgradeable high-quality IP Phone that is unparalleled in features, value,  
and support.  
SPA922  
Based on the SIP standard, the SPA922 has been tested to ensure comprehensive  
interoperability with equipment from VoIP infrastructure leaders, enabling service  
providers to quickly roll out competitive, feature-rich services to their customers.  
With hundreds of features and configurable service parameters, the SPA922 addresses  
the requirements of traditional business users while leveraging the advantages of IP  
telephony. Features such as easy station moves, presence, and shared line appearances  
(across local and geographically-dispersed locations) are just some of the many  
advantages.  
Standard features on the SPA922 include dual-switched Ethernet ports, 802.3af Power over  
Ethernet (POE)* support, a high-resolution graphical display, full-duplex speakerphone,  
and a 2.5 mm headset port. The SPA922 supports one line with two call appearances and  
provides support for three-way conferencing, attended call transfer, and placing a call on  
hold to answer an incoming call. The line can be configured as a unique phone number (or  
extension), or can be configured to share a number that is assigned to multiple phones.  
The SPA922 uses standard encryption protocols to provide secure remote provisioning and  
unobtrusive in-service software upgrades. Linksys secure remote provisioning tools include  
detailed performance measurement and troubleshooting features, enabling network providers  
to deliver high-quality support to their subscribers. Remote provisioning also saves service  
providers the hassle and expense of managing, preloading, and re-configuring customer premise  
equipment (CPE).  
*The power supply for the SPA922 is sold separately and is required if PoE functionality is not implemented.  
Full-featured one-line  
business-class IP phone  
supporting Power over  
Ethernet 802.3af  
Connect directly to an  
Internet Telephone Service  
Provider or connect to an  
IP PBX  
Dual-switched Ethernet  
ports, full-duplex  
speakerphone, caller ID,  
call hold, conferencing, and  
more  
Easy installation and secure  
remote provisioning—  
menu-based and web-  
based configuration  
PRODUCT DATA  
BUSINESS SERIES  
IP Phone  
Model: SPA922  
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Functions and Features  
One Voice Line with Two Call Appearances  
Line Status—Active Line Indication, Name and Number  
Menu-Driven User Interface—Multiple Languages Supported  
Digits Dialed with Number Auto-Completion  
Shared Line Appearance**  
Full-Duplex Speakerphone  
Call Hold  
Music on Hold**  
Call Waiting  
Caller ID Name and Number  
Outbound Caller ID Blocking  
Call Transfer—Attended and Blind  
Three-Way Call Conferencing with Local Mixing  
Connects to External Conference Bridge for Multi-party Conferencing  
Automatic Redial of Last Calling and Last Called Numbers  
On-Hook Dialing  
Call Pick Up—Selective and Group**  
Call Park and Retrieval**  
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  
Redial from Call Logs  
Personal Directory with Auto-dial (100 entries)  
Do Not Disturb (callers hear line busy tone)  
URI (IP) Dialing Support (Vanity Numbers)  
On-Hook Default Audio Configuration (Speakerphone and Headset)  
Multiple Ring Tones with Selectable Ring Tone per Line  
Called Number with Directory Name Matching  
Call Number using Name—Directory Matching or via Caller ID  
Subsequent Incoming Calls with Calling Name and Number  
Date and Time with Intelligent Daylight Savings Support  
Call Duration and Start Time Stored in Call Logs  
Call Timer  
Name and Identity (Text) Displayed at Start Up  
Distinctive Ringing Based on Calling and Called Number  
Speed Dialing—8 entries  
Configurable Dial/Numbering Plan Support  
Intercom** and Group Paging**  
NAT Traversal, including STUN Support  
DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy  
Syslog and Debug Server Records (Configurable Per Line)  
Report Generation and Event Logging  
PRODUCT DATA  
BUSINESS SERIES  
Statistics Transmitted in BYE Message  
Secure Call Encrypted Voice Communication Support - SIP over TLS, and SRTP  
Built-in Web Server for Administration and Configuration with Multiple Security Levels  
Automated Provisioning, Multiple Methods—up to 256 Bit Encryption (HTTP, HTTPS, TFTP)  
Asynchronous Notification of Upgrade Availability via NOTIFY  
Non-intrusive, In-Service Upgrades  
Optionally Require Admin Password to Reset Unit to factory Defaults  
** Feature requires support by SIP server  
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Data Networking  
MAC Address  
(IEEE 802.3)  
IPv4  
ARP  
Internet Protocol v4 (RFC 791), upgradeable to v6 (RFC 1883)  
Address Resolution Protocol  
DNS  
DHCP Client  
ICMP  
TCP  
UDP  
RTP  
RTCP  
DiffServ  
A Record (RFC 1706), SRV Record (RFC 2782)  
Dynamic Host Configuration Protocol (RFC 2131)  
Internet Control Message Protocol (RFC792)  
Transmission Control Protocol (RFC793)  
User Datagram Protocol (RFC768)  
Real Time Protocol (RFC 1889) (RFC 1890)  
Real Time Control Protocol (RFC 1889)  
(RFC 2475)  
Type of Service-TOS  
VLAN Tagging 802.1p/q  
SNTP  
(RFC 791/1349)  
Layer 2 QoS  
Simple Network Time Protocol (RFC 2030)  
Voice  
SIPv2—Session Initiation Protocol Version 2 (RFC 3261, 3262, 3263, 3264)  
SIP Proxy Redundancy—Dynamic via DNS SRV, A Records  
Re-registration with Primary SIP Proxy Server  
SIP Support in Network Address Translation Networks—NAT (including STUN)  
SIPFrag (RFC 3420)  
Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP  
Codec Name Assignment  
Voice Algorithms:  
G.711 (A-law and μ-law, G.726 (16/24/32/40 kbps), G.729 A, G.723.1 (6.3 kbps, 5.3 kbps)  
Dynamic Payload Support  
Adjustable Audio Frames Per Packet  
DTMF—In-band and Out-of-Band (RFC 2833) (SIP INFO)  
Flexible Dial Plan Support with Inter-Digit Timers  
IP Address/URI Dialing Support  
Call Progress Tone Generation  
Jitter Buffer—Adaptive  
Frame Loss Concealment  
VAD—Voice Activity Detection with Silence Suppression  
Attenuation/Gain Adjustments  
Linksys  
A Division of Cisco Systems, Inc.  
121 Theory  
Irvine, CA 92617 USA  
E-mail:  
Web:  
MWI—Message Waiting Indicator Tones  
VMWI—Voice Mail Waiting Indicator—Via NOTIFY, SUBSCRIBE  
Caller ID Support (Name and Number)  
Linksys products are available in more  
than 50 countries, supported by  
12 Linksys Regional Offices throughout  
the world. For a complete list of local  
Linksys Sales and Technical Support  
contacts, visit our worldwide website  
Third Party Call Control (RFC 3725)  
Linksys Phone Comparison Chart  
Model  
Voice Lines  
Ethernet  
Ports  
High-Resolution  
Graphical Display  
Power over Ethernet  
Support  
SPA901  
SPA921  
SPA922  
SPA941  
SPA942  
SPA962  
1
1
1
2
1
2
2
No  
No  
No  
Yes  
No  
Yes  
Yes  
PRODUCT DATA  
BUSINESS SERIES  
1
Yes  
1
Yes  
2 to 4  
2 to 4  
6
Yes  
Yes  
Yes, Color  
Check the product package and contents for specific features supported. Specifications are subject to change without notice.  
Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries.  
Copyright © 2008 Cisco Systems, Inc. All rights reserved. Other brands and product names are trademarks or registered trademarks of their  
respective holders.  
8022020NC-KO  
Model: SPA922  
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