Cisco SPA942 4-line IP Phone with 2-port Switch Description

Cisco SPA942 4-Line IP Phone with 2-Port Switch
Stylish and functional in design, the Cisco 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 SPA942 uses industry-leading voice over IP (VoIP) technology from Cisco to deliver an upgradeable high-quality IP phone that is unparalleled in features, value, and support. Based on the Session Initiation Protocol (SIP), the Cisco SPA942 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 Cisco SPA942 addresses the requirements of traditional business users while taking advantage of the benefits of IP telephony. 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 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.

Cisco SPA942 4-line IP Phone with 2-port Switch 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

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