Cisco SPA303 VoIP/SIP Phone, 3-Line, LCD Display

Cisco SPA303

For small businesses that need affordable, reliable, and easy-to-use IP phones, the Cisco SPA303 3-Line IP Phone with Ethernet Ports and LCD display can be used with either Session Initiation Protocol (SIP) or Smart Phone Control Protocol (SPCP) and is supported with the Cisco Unified Communications 500 Series, Cisco SPA9000 Voice System, and hosted IP telephony systems. The SPA303 enables many productivity-enhancing features and XML applications. It is easy to install and supports highly secure remote provisioning as well as web-based configuration.

spa303

Key Features

  • 3-line business-class IP phone.
  • Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX).
  • Dual switched Ethernet ports, speakerphone, caller ID, call hold, conferencing, and more.
  • Easy installation and highly secure remote provisioning, as well as menu-based and web-based configuration.
  • Supports both Session Initiation Protocol (SIP) and Smart Phone Control Protocol

Specification

Telephony Features
  • 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 dialling
  • 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 dialled with number auto-completion
  • Anonymous caller blocking
  • Support for Uniform Resource Identifier (URI) (IP) dialling (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 start-up
  • Distinctive ringing based on calling and called number
  • 10 user-downloadable ring tones
  • Speed dialling, 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

 

Hardware Features

 

  • Pixel-based display: 128 x 64 monochrome LCD graphical display
  • Dedicated illuminated buttons for:
    • Audio mute on/off
    • Headset on/off
    • Speakerphone on/off
  • Four-way rocking directional knob for menu navigation
  • Voicemail message waiting indicator light
  • Voicemail message retrieval button
  • Dedicated hold button
  • Settings button for access to feature, setup, and configuration menus
  • Volume control rocking up/down knob controls handset, headset, speaker, ringer
  • Standard 12-button dialling pad
  • High-quality handset and cradle
  • Built-in high-quality microphone and speaker
  • Headset jack: 2.5 mm
  • LED test function
  • Two Ethernet LAN ports with integrated Ethernet switch: 10/100BASE-T RJ-45
  • 5 VDC universal (100-240V) switching included
Security Features

 

  • Password-protected system, preset to factory defaults
  • Password-protected access to administrator and user-level features
  • HTTPS with factory-installed client certificate
  • HTTP digest: encrypted authentication via MD5 (RFC 1321)
  • Up to 256-bit Advanced Encryption Standard (AES) encryption

 

Data Networking

 

  • MAC address (IEEE 802.3)
  • IPv4 (RFC 791)
  • Address Resolution Protocol (ARP)
  • DNS: A record (RFC 1706), SRV record (RFC 2782)
  • Dynamic Host Configuration Protocol (DHCP) client (RFC 2131)
  • Internet Control Message Protocol (ICMP) (RFC 792)
  • TCP (RFC 793)
  • User Datagram Protocol UDP (RFC 768)
  • Real Time Protocol RTP (RFC 1889, 1890)
  • Real Time Control Protocol (RTCP) (RFC 1889)
  • Real Time Control Protocol – Extended Report ( RTCP-XR) ( RFC 3611 )
  • Differentiated Services (DiffServ) (RFC 2475)
  • Type of service (ToS) (RFC 791, 1349)
  • VLAN tagging 802.1p/Q: Layer 2 quality of service (QoS)
  • Simple Network Time Protocol (SNTP) (RFC 2030)

 

Dimensions (W x H x D)
  • 8.66 x 7.80. x 1.18 in. (220 x 198 x 30 mm)
Voice Gateway

 

  • SIP version 2 (RFC 3261, 3262, 3263, 3264)
  • SPCP with the Cisco Unified Communications 500 Series
  • SIP proxy redundancy: dynamic via DNS SRV, A records
  • Re-registration with primary SIP proxy server
  • SIP support in NAT networks (including STUN)
  • SIPFrag (RFC 3420)
  • Highly secure (encrypted) calling via Secure Real-Time Transport Protocol (SRTP)
  • SIP/TLS
  • Codec name assignment
  • Voice algorithms:
    • G.711 (A-law and ยต-law)
    • G.726 (16/24/32/40 kbps)
    • G.729 AB
    • G.722
  • Dynamic payload support
  • Adjustable audio frames per packet
  • Dual-tone multifrequency (DTMF), in-band and out-of-band (RFC 2833) (SIP INFO)
  • Flexible dial plan support with interdigit timers
  • IP address/URI dialling support
  • Call progress tone generation
  • Jitter buffer: adaptive
  • Frame loss concealment
  • Voice activity detection (VAD) with silence suppression
  • Attenuation/gain adjustments
  • Message waiting indicator (MWI) tones
  • Voicemail waiting indicator (VMWI), via NOTIFY, SUBSCRIBE
  • Caller ID support (name and number)
  • Third-party call control (RFC 3725)