Cisco SPA 301 IP Phone Kenya

The Cisco SPA 301 IP Phone in Kenya is an inexpensive 1-line SIP-based entry level phone for business or home office use. The Cisco SPA301 Kenya IP Phone has been verified to confirm full interoperability. The SPA 301 IP Phone Kenya performs highly safe remote provisioning and modest in-service software upgrades using standard encryption protocols. Remote provisioning tools contain full performance measurement and troubleshooting features, allowing network providers to provide high-quality support to their subscribers.

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Description

Cisco SPA 301 IP Phone Mombasa

The Cisco SPA 301 IP Phone in Mombasa is not only inexpensively perfect for small budgets but it’s a completely featured basic internet phone. Full featured and basic – Isn’t that a flaw. It’s the simplicity of the Cisco SPA 301 IP Phone that delivers its important operation. But in its flexibility to the increasing host of features and functions offered by service providers you’ll find an instinctive and satisfying user experience that’s never lacking.

Cisco IP phones Kenya are designed to stay on the cutting edge. Flexibility to either connects the Cisco SPA301 IP Phone Kenya directly to an internet phone service provider or a more hosted system. Powerful features at your command that naturally work with you. Perfect for any small office, Cisco SPA301 IP phone is affordable today and future proof for tomorrow.

For small businesses that need cheap, consistent, and easy-to-use Cisco phones, the Cisco SPA301 IP Phone is a 1-Line IP Phone which can be used with either SIP or Smart Phone Control Protocol and is supported with the Cisco Unified Communications 500 Series, Cisco SPA 9000 Voice System and hosted IP telephony systems. The Cisco SPA301 IP Phone is easy to install and supports highly safe remote provisioning as well as web-based configuration. It addresses the desires of the traditional business user while building on the benefits of IP telephony.

Cisco SPA301 IP Phone 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
One service provider line
Two call appearances accessed via flash key or hook flash
Shared line appearance **
Line status indicator

Ultra Clear Backlit Graphical Display Screen
- The Cisco SPA301 IP Phone boasts of a Grayscale 128 x 64 – pixel – based monochrome LCD graphical display for an enhanced visual clarity. The display screen also gives the user a quick access to details since it has programmable backlit line keys and interactive soft buttons that guide you through all call features and functions.
HD Audio
- The Cisco SPA301 IP Phone boasts of a built-in full-duplex speakerphone which supports the use of wideband audio codecs to deliver crystal clear sounds, ensuring that both the listener and the speaker can enjoy clear, natural and life-like conversations. For added security, the audible Dual Tone Multi-frequency (DTMF) tones are masked when the speakerphone mode is used.
Easy User Interface
- Cisco SPA301 IP Phone is an easy-to-use gadget which requires no technical expertise of its user. It is equipped with permanently-labeled buttons which are used to access various functions such as Speaker, Headset, Mute, Volume, Phone, History, Contacts, and Voicemail Message, as well as a permanently-labeled 4-way navigation cluster which enables the user to navigate the screen easily as desired. This saves time therefore maximizing efficiency in the office.
Headset Support
- The Cisco SPA301 IP Phone comes with an integrated headset port with HD Voice support which allows you to directly connect your favorite headset with compatible cable or answer your cordless headset away from your desk with use of a compatible EHS cable. This feature enables you to multitask in the office, thus saving time and maximizing productivity. This also eliminates the need for a separate headset amplifier and allows the handset to remain in its cradle, making headset use simpler.
Supports Class 1 Power over Ethernet (PoE)
- The Cisco SPA301 IP Phone is comes with dual 10/100Mbps Ethernet switch ports which are used to connect the IP phone to a Power over Ethernet.
Wall Mountable Device
- The Cisco SPA301 IP phone offers an easy installation process. It can be mounted on a wall to achieve a cleaner workstation using an included dual – angle stand. The stand is adjustable to two different angles(supports viewing angles of 30 degrees and 45 degrees) to provide optimum display viewing and comfortable use of all buttons and keys. The foot-stand is keyed to match standard wall-jack configurations for wall mounting. Optional wall-mount brackets are also offered.
Carrier-Grade Security, Provisioning, and Management
- The Cisco SPA 301 IP Phone 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.

Cisco SPA 301 IP Phone Technical Specifications

Telephony Features

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
Hotline and warm line automatic calling
Digits dialled with number auto-completion
Anonymous caller blocking
Support for Uniform Resource Identifier (URI) (IP) dialling (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 ringtones
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

Voicemail message waiting indicator light
Voicemail message retrieval button
Volume control
Redial Button
Flash Button
Standard 12-button dialling pad
High-quality handset
One Ethernet WAN port 10/100BASE-T RJ-45
5 VDC universal (100-240V) switching power adaptor included

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
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 µ-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 dialing 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)