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Cisco SPA301-G1 1-Line IP Phone

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Highlights:

A SPA 301 Cisco VoIP Phone has full standards-based SIP implementation, giving it wide compatibility with a range of VoIP services, both locally-hosted and offsite. Full encryption allows for secure remote provisioning, as well as unobtrusive live software updates. The SPA 301 is fully configurable via web GUI, with built-in diagnostics, call time logging, and troubleshooting features.

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MFR # SPA301-G1 SKU: SPA301-G1 Category: Tag: Brand:
Description

Cisco SPA 301 1-Line IP Phone

  •  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
Additional information
Weight 2 lbs
Dimensions 11.5 × 8 × 2.75 in
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 (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 Series, SIP proxy redundancy: dynamic via DNS SRV, A records, Reregistration with primary SIP proxy server, SIP support in NAT networks (including STUN), SIPFrag (RFC 3420), Secure (encrypted) calling via SRTP, Codec name assignment, Voice algorithms:, G.711 (A-law and -law), G.726 (16/24/32/40 kbps), G.729 A, 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 dialing support, Call progress tone generation, Jitter buffer: adaptive, Frame loss concealment, Comfort Noise Generation (CNG), Voice activity detection (VAD) with silence suppression, Attenuation/gain adjustments, VMWI – Voicemail waiting indicator, via NOTIFY, SUBSCRIBE, Third-party call control (RFC 3725)

Administration and Maintenance

Integrated web server provides web-based administration and configuration, Automated provisioning and upgrade via TFTP, HTTP or HTTPS, Asynchronous notification of upgrade availability via NOTIFY, Nonintrusive in-service upgrades, Report generation and event logging, Statistics transmitted in BYE message, RTCP-XR, Syslog and debug server records: configurable per line

Physical Interfaces

One 10/100BASE-T RJ-45 Ethernet port (IEEE 802.3), Handset: RJ-9 connector

Power Supply

Switching type (100-240V) automatic, DC input voltage: +5 VDC at 1.0A maximum

Indicators

Message waiting indicator LED

Operating Temperature

32 ~ 113ºF (0 ~ 40ºC)

Storage Temperature

-13 ~ 185ºF (-20 ~ 70ºC)

Operating Humidity

5 to 95% non condensing

Storage Humidity

5 to 95% non condensing

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