SPA301 Cisco Small Business 1-Line SIP Phone (SPA301) | Like New
SKU: SPA301-G1-LN
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Cisco SPA301 IP Phone
Overview:
Comprehensive Interoperability and SIP-Based Feature Set The Cisco SPA 301 1-Line IP Phone 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 301 addresses the requirements of traditional business users while building on the advantages of IP telephony. Features such as easy station moves and shared line appearance (across local and geographically dispersed locations) are just some of the many advantages of the SPA 301. Carrier-Grade Security, Provisioning, and Management The Cisco SPA 301 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.
Features:
Basic 1-line business-class IP phone Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)
Easy installation and secure remote provisioning, as well as web-based configuration Supports both Session Initiation Protocol (SIP) and Smart Phone Control Protocol (SPCP) with the Cisco Unified Communications 500 Series One voice line Music on hold*
Call waiting Outbound caller ID blocking** Call transfer: attended and blind** ( SIP Only )
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 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 Digits dialed with number auto-completion
Anonymous caller blocking**
Support for Uniform Resource Identifier (URI) (IP) dialing (vanity numbers)
Multiple ring tones with selectable ring tone per line Call duration and start time stored in call logs in web GUI
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 *Feature requires support by call server**Feature activated via feature code Part Number: SPA301-G1
Specifications
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)Voice gateway
SIP version 2 (RFC 3261, 3262, 3263, 3264)SPCP with the Cisco Unified Communications 500 SeriesSIP proxy redundancy: dynamic via DNS SRV, A recordsRe-registration with primary SIP proxy serverSIP support in NAT networks (including STUN)SIPFrag (RFC 3420)Highly secure (encrypted) calling via Secure Real-Time Transport Protocol (SRTP)SIP/TLSCodec name assignmentVoice algorithms:G.711 (A-law and ?-law)G.726 (16/24/32/40 kbps)G.729 ABG.722Dynamic payload supportAdjustable audio frames per packetDual-tone multifrequency (DTMF), in-band and out-of-band (RFC 2833) (SIP INFO)Flexible dial plan support with interdigit timersIP address/URI dialing supportCall progress tone generationJitter buffer: adaptiveFrame loss concealmentVoice activity detection (VAD) with silence suppressionAttenuation/gain adjustmentsMessage waiting indicator (MWI) tonesVoicemail waiting indicator (VMWI), via NOTIFY, SUBSCRIBECaller ID support (name and number)Third-party call control (RFC 3725)Provisioning, administration,
and maintenanceIntegrated web server provides web-based administration and configurationTelephone keypad configuration via display menu/navigationAutomated provisioning and upgrade via HTTPS, HTTP, TFTPAsynchronous notification of upgrade availability via NOTIFYNonintrusive in-service upgradesReport generation and event loggingStatistics transmitted in BYE messageRTCP-XRSyslog and debug server records: configurable per linePower supply
Switching type (100-240V) automaticDC input voltage: +5 VDC at 1.0A maximumPhysical interfaces
Two 10/100BASE-T RJ-45 Ethernet ports (IEEE 802.3)Handset: RJ-9 connectorBuilt-in speakerphone and microphoneHeadset 2.5-mm portIndicator lights/LED
Speakerphone on/off button with LEDHeadset on/off button with LEDMute button with LEDMessage waiting indicator LEDLED test functionDimensions (W x H x D)
8.66 x 7.80. x 1.18 in. (220 x 198 x 30 mm)
Unit weight
1.50 lb ( 0.68kg)
Operating temperature
32º ~ 113ºF (0º ~ 40ºC)
Storage temperature
-13º ~ 185ºF (-20º ~ 70ºC)
Operating humidity
5% to 95% noncondensing
Storage humidity
5% to 95% noncondensing