Cisco Linksys SPA941 4-line IP Phone with 1-port Ethernet

Product Description

Four lines of Internet telephony are at your fingertips. Formerly part of the Linksys Voice System (LVS), now part of Cisco Small Business IP Phone Series, the SPA941 can be connected directly to an Internet telephone service provider or to an IP PBX via its single Ethernet port. 

Each line can be independently assigned its own phone number or extension. A high-resolution graphical display makes for easy menu- and web-based configuration.

Additional features of the Cisco SPA941 IP Phone include: 

Speakerphone, caller ID, call hold, transfer, conferencing, and more
Easy installation and secure remote provisioning
A high-resolution graphical display, speakerphone, and a 2.5 mm headset port
Features such as easy station moves, presence, and shared line appearances
Standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades
Detailed performance measurement and troubleshooting features that help network providers deliver high-quality support

Data networking

• MAC address (IEEE 802.3)
• IPv4 - Internet Protocol v4 (RFC 791)
• ARP - Address Resolution Protocol
• DNS - A record (RFC 1706), SRV record (RFC 2782)
• DHCP client - Dynamic Host Configuration Protocol (RFC 2131)
• ICMP - Internet Control Message Protocol (RFC 792)
• TCP - Transmission Control Protocol (RFC 793)
• UDP - User Datagram Protocol (RFC 768)
• RTP - Real Time Protocol (RFC 1889, 1890)
• RTCP - Real Time Control Protocol (RFC 1889)
• DiffServ - Differentiated Services (RFC 2475)
• ToS - Type of service (RFC 791, 1349)
• VLAN tagging 802.1p/Q - Layer 2 quality of service (QoS)
• SNTP - Simple Network Time Protocol (RFC 2030)

Voice gateway

• SIP version 2 (RFC 3261, 3262, 3263, 3264)
• 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 prestandard implementation of Secure RTP
• Codec name assignment
• Voice algorithms:
• G.711 (A-law and μ-law)
• G.726 (16/24/32/40 kbps)
• G.729 A
• G.723.1 (6.3 kbps, 5.3 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
• Frame loss concealment
• Voice activity detection (VAD) with silence suppression
• Attenuation/gain adjustments
• 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 for 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
• Syslog and debug server records: configurable per line

Physical Interfaces

• One 10BASE-T RJ-45 Ethernet port (IEEE 802.3)
• Handset: RJ-9 connector
• Built-in speakerphone and microphone
• Headset 2.5-mm port

Power supply

• Switching type (100-240V) automatic
• DC input voltage: +5 VDC at 2.0A maximum
• Power adapter: 100-240V 50-60 Hz (26-34 VA) AC input

Indicator lights/LED

• 4 line key buttons with LED
• Speakerphone on/off button with LED
• Headset on/off button with LED
• Mute button with LED
• Message waiting indicator LED

Dimensions (W x H x D)

• 7.68 x 6.30. x 7.09 in. (195 x 160 x 180 mm)

Unit weight

• 2.15 lb ( 0.9752 kg)

Operating temperature

• 41º ~ 113ºF (5º ~ 45ºC)

Storage temperature

• 13º ~ 185ºF (-25º ~ 85ºC)

Operating humidity

• 10% to 90% noncondensing

Storage humidity

• 10% to 90% noncondensing