VoIP телефон Cisco SPA508G в пакет със слушалка с микрофон Jabra BIZ 1500 Mono, 128x64 монохромен LCD дисплей, 8 линии, 10/100BASE-T & PoE, бързи бутони, TFTP протокол, черен

Cisco SPA508G + Jabra BIZ 1500 Mono

Highlights
• Full-featured 8-line business-class IP phone supporting Power over Ethernet (PoE)
• Monochrome backlit display for ease of use, aesthetics, and on-screen applications
• Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)
• Wideband audio for unsurpassed voice clarity and enhanced speaker quality
• Easy installation and highly secure remote provisioning, as well as menu-based and web-based configuration
• Supports up to two Cisco® SP500S Expansion Module, adding up to 64 additional buttons*
• Supports both Session Initiation Protocol (SIP) and Smart Phone Control Protocol (SPCP) with the Cisco Unified Communications 500 Series for Small Business

Comprehensive Interoperability and SIP-Based Feature Set
Part of the Cisco Small Business Pro Series, the SIP-based Cisco SPA508G 8-Line IP Phone (Figure 1) has been tested to 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 SPA508G 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 SPA508G.

The Cisco SPA508G 8-Line IP Phone also supports productivity-enhancing features such as VoiceView Express, and Cisco XML applications when used with the Cisco Unified Communications 500 Series in SPCP mode.
Carrier-Grade Security, Provisioning, and Management

The Cisco SPA508G 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.

Telephony Features
• Eight voice lines
• Eight independent SIP Registrations*
• Line status: active line indication, with name and number
• Menu-driven user interface
• Shared line appearance**
• Speakerphone
• Call hold
• Music on hold**
• Call waiting
• 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
• On-hook dialing
• Call pickup: selective and group**
• Call park and unpark**
• Call swap
• Call back on busy
• Call blocking: anonymous and selective
• Call forwarding: unconditional, no answer, on busy
• Hot line and warm line automatic calling
• Call logs (60 entries each): made, answered, and missed calls
• Redial from call logs
• Personal directory with auto-dial (100 entries)
• Do not disturb
• Digits dialed with number auto-completion
• Anonymous caller blocking
• Uniform Resource Identifier (URI) (IP) dialing support (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 start time stored in call logs
• Call timer
• 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
• Intercom**
• Group paging**
• Network Address Translation (NAT) Traversal, including Simple Traversal of UDP Through NATs (STUN) support
• DNS SRV and multiple A records for proxy lookup and proxy redundancy
• Syslog, debug, report generation, and event logging
• Highly secure call encrypted voice communications support
• 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

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 Transport Protocol (RTP) (RFC 1889, 1890)
• Real Time Control Protocol (RTCP) (RFC 1889)
• 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)
• Cisco Discovery Protocol (CDP)
• Link Layer Discovery Protocol (LLDP)

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 μ-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
• VMWIVoicemail waiting indicator, 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
• Syslog and debug server records: configurable per line

Power supply
• Power supply is optional and is purchased separately
• Models: Cisco PA100-NA, PA100-EU, PA100-UK, PA100-AU
• DC output voltage: +5 VDC at 2.0A maximum
• Switching power adapter: 100-240V 50-60 Hz AC input

Physical interfaces
• Two 10/100BASE-T RJ-45 Ethernet ports (IEEE 802.3)
• Handset: RJ-9 connector
• Built-in speakerphone and microphone
• Headset 2.5mm jack

Indicator lights/LEDs
• Speakerphone on/off button with LED
• Headset on/off button with LED
• Mute button with LED
• Message waiting LED

Body dimensions (W x H x D)
8.42 x 8.35. x 1.73in. (214 x 212 x 44 mm)
Unit weight
2.00 lb ( 0.9 kg)

Operating temperature
32º ~ 104ºF (0º ~ 40ºC)

Storage temperature
-4º ~ 158ºF (-20º ~ 70ºC)

Operating humidity
5% to 95% noncondensing

Storage humidity
5% to 95% noncondensing



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