Cisco SPA303 Unified IP Phone

Cisco SPA303 Unified IP Phone

Comprehensive Interoperability and SIP-Based Feature Set

Based on SIP, the Cisco SPA 303 3-Line IP Phone with 2-Port Switch has been tested to help ensure comprehensive interoperability with equipment from voice over IP (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 SPA303 addresses the requirements of traditional business users while building on the advantages 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 of the Cisco SPA303. The Cisco SPA 303 IP phone can also be used with productivity-enhancing features such as VoiceView Express, and Cisco XML applications when interfacing with the Cisco Unified Communications 500 Series in SPCP mode.

Carrier-Grade Security, Provisioning, and Management

The Cisco SPA303 uses standard encryption protocols to perform highly secure remote provisioning and unobtrusive in-service software upgrades. 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 time and expense of managing, preloading, and reconfiguring customer premises equipment.

Specifications

• Three voice lines
• Pixel-based display: 128 x 64 monochrome graphical liquid crystal display (LCD)
• Line status: active line indication, name and number
• Menu-driven user interface
• Shared line appearance*
• 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
• Multiparty conferencing via external conference bridge
• Automatic redial of last calling and last called numbers
• On-hook dialing
• Call pickup: selective and group*
• Call park and unpark*
• Call swap
• Call back on busy**
• Call blocking: anonymous and selective
• Call forwarding: unconditional, no answer, and 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
• Digits dialed with number auto-completion
• Anonymous caller blocking
• Support for Uniform Resource Identifier (URI) (IP) dialing (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
• Ability to call number using name: directory matching or via caller ID
• Subsequent incoming calls show calling name and number
• Date and time with support for intelligent daylight savings
• 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
• 10 user-downloadable ring tones
• Speed dialing, eight entries
• Configurable dial/numbering plan support
• Intercom*
• Group paging
• Network Address Translation (NAT) traversal, including Serial Tunnel (STUN) support
• DNS SRV and multiple A records for proxy lookup and proxy redundancy
• Syslog, debug, report generation, and event logging
• Support for highly secure encrypted voice communications
• Built-in web server for administration and configuration with multiple security levels
• Automated remote provisioning, multiple methods; up to 256 bit encryption (HTTP, HTTPS, Trivial File Transfer Protocol [TFTP])
• Option to require administrator password to reset unit to factory defaults