Cisco Small Business SPA303G IP Phones for Home Office
Cisco SPA303G is Basic and Affordable IP Phone for Business or Home Office
• 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 (SPCP) with the Cisco Unified Communications 500 Series 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 SPA 303 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 SPA 303. 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 SPA 303 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. • 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* • 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 • Call pickup: selective and group* • 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 • Personal directory with auto-dial (100 entries) • 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 • 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 • 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
**Feature requires support by call server.**Feature activated via feature code. • Pixel-based display: 128 x 64 monochrome LCD graphical display • Dedicated illuminated buttons for: • Four-way rocking directional knob for menu navigation • Voicemail message waiting indicator light • Voicemail message retrieval 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 dialing pad • High-quality handset and cradle • Built-in high-quality microphone and speaker • Two Ethernet LAN ports with integrated Ethernet switch: 10/100BASE-T RJ-45 • 5 VDC universal (100-240V) switching included • FCC (Part 15, Class B) , UL, CE Mark, A-Tick • 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 • Quick-start installation and configuration guide • Provisioning guide (for service providers only) • Cisco SPA 303 IP phone, handset, and stand • Quick installation guide
Specifications for the Cisco SPA 303- 3 Line IP Phone
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| • MAC address (IEEE 802.3) • 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) • 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) |
| • 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) • Highly secure (encrypted) calling via Secure Real-Time Transport Protocol (SRTP) • G.711 (A-law and μ-law)• G.726 (16/24/32/40 kbps)• 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 dialing support • Call progress tone generation • Jitter buffer: adaptive • 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) |
Provisioning, administration,
and maintenance | • Integrated web server provides web-based administration and configuration • Telephone keypad configuration via display menu/navigation • Automated provisioning and upgrade via HTTPS, HTTP, TFTP • Asynchronous notification of upgrade availability via NOTIFY • Nonintrusive in-service upgrades • Report generation and event logging • Statistics transmitted in BYE message
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