Stylish and functional in design, the Cisco - Linksys SPA942 4-Line IP Phone with 2-Port Switch is ideal for a residence or business using a hosted IP telephony service, an IP private branch exchange (PBX), or a large-scale IP Centrex deployment. The Cisco - Linksys SPA942 uses industryleading voice over IP (VoIP) technology from Cisco to deliver an upgradeable high-quality IP phone that is unparalleled in features, value, and support.
Features such as easy station moves and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages. Standard features on the Cisco - Linksys SPA942 include four active lines, 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. Each line can be independently configured to use a unique phone number (or extension), or can use a shared number that is assigned to multiple phones.
The Cisco - Linksys SPA942 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Highly 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 reconfiguring customer premises equipment (CPE).
*The power supply for the SPA942 is sold separately and is required if PoE capability is not implemented.
Cisco Linksys SPA942 Highlights
- Full-featured 4-line business-class IP phone supporting Power over Ethernet 802.3af
- Dual switched Ethernet ports, speakerphone, caller ID, call hold, conferencing, and more
- Monochrome LCD graphical display with backlight
- Connects directly to an Internet telephone service provider or to an IP PBX
- Support SIP-compliant IP PBX such as Asterisk
- Supports advanced features and automatic provisioning with the Cisco - Linksys SPA9000 Small Business Voice System
- Support for Call Ctrl Call Manager and brings the SPA942 to Windows multiplying the SPA942 capabilities. Call Ctrl opens all phone features from Windows and applications users use daily. Features such as Click-to-Dial from Outlook, Internet Explorer and essentially any Windows application are now available to the Windows desktop.
Windows and desktop application integration
Call Ctrl brings the functionality of your SPA942 to your desktop. This is the only solution currently available and developed specifically for SPA942. Call Ctrl was developed to fill the gap between the phone and your computer. Binding the phone and desktop together multiplies the SPA942 capabilities which exponentially increases both productivity and features.
Dialing calls from most Windows applications such as Outlook or Internet Explorer is made possible through Call Ctrl. After Call Ctrl is installed a simple wizard-style setup is used to connect Call Ctrl to the SPA942. Phone functions such as making calls, hang up, hold, mute, transfer, conference are all available from within Windows. A pop-up or screen-pop is made for new incoming calls where a user can perform a number of functions to properly handle the incoming call. Call Ctrl reports on all phone number activity. Making these features available in Windows empower users with increased productivity.
- Perform all phone functions from their Windows desktop and the applications they use daily
- Shared phone numbers between users
- Click-to-Dial from Outlook and Internet Explorer
- Dial calls from essentially any Windows application
- Phone Number and Contact Management
Cisco - Linksys SPA942 Features
- Up to four lines with independent configuration and registration
- Active line indication, with name and number
- Menu-driven user interface, with support for multiple languages
- 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
- Call conferencing
- Automatic redial
- On-hook dialing
- Call pickup - selective and group**
- Call park and retrieval**
- Call swap
- Call back on busy
- Call blocking - anonymous and selective
- Call forwarding - unconditional, no answer, or busy
- Hot line and warm line automatic calling
- Call logs (60 entries each) - calls made, answered, and missed
- Redial from call logs
- Personal directory with auto-dial (100 entries)
- Do not disturb (callers hear busy signal)
- Uniform Resource Identifier (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 startup
- Distinctive ringing based on calling and called number
- Speed dialing
- Configurable dial/numbering plan support (per line)
- Intercom** and group paging**
- 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
- Statistics transmitted in BYE message
- Secure call encrypted voice communication support - SIP over Transport Layer Security (TLS), and Secure Real-Time Transport Protocol (SRTP)
- Built-in web server for administration and configuration with multiple security levels
- Automated provisioning, multiple methods - up to 256 bit encryption (HTTP, HTTPS, Trivial File Transfer Protocol [TFTP])
- Asynchronous notification of upgrade availability via NOTIFY
- Nonintrusive, in-service upgrades
- Optionally require administrator password to reset unit to factory defaults
** Feature requires support by SIP server