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Descripción del producto:
Teléfono Cisco SPA508G
Posee 8 líneas IP
Tiene 2 puertos Gigabit Ethernet
Puede ser alimentado a través de PoE o con la utilización de un eliminador de energía (el cual se adquiere por separado).
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Compatibilidad:
- Centrales de comunicación PBX tipo IP SIP tales como: Asterisk, Elastix, FreeSwitch, OpenPBX, FreePBX, entre otros.
- Cisco Unified Communication 500 Series
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Estado:
-El equipo ha sido retirado de un ambiente de producción por reemplazo; posee señales de uso, puede tener raspones simples o manchas. Las fotos aquí mostradas son reales del teléfono.
-El aparato posee hardware 100% funcional. No presenta daños que impida su funcionalidad. Previamente ha sido limpiado y verificado.
-Posee el último firmware otorgado por el fabricante (al momento de su retiro de producción)
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Contenido del empaque:
- Cisco SPA508G IP Phone
- Handset
- Stand
- Handset cord
No contiene:
- Cable Ethernet RJ-45
- Guia de instalación rápida
- CD o Documentación
- Empaques originales
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Garantía:
- 30 días naturales una vez realizada la compra
- Compra protegida por mercado libre.
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Notas importantes:
-Tomar en cuenta que el equipo ya no se comercializa por el fabricante. Ya no se puede adquirir soporte, garantía y/o servicios por parte de Cisco.
- No se ofrece factura. De requerirla háznoslo saber para hacerte una publicación especial que incluya los impuestos respectivos.
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Envío:
Envío gratis a todo el país a través de paquetería de Mercado Libre.
El producto se embarca un día después de haber recibido confirmación de tu pago.
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Especificaciones:
Cisco Small Business IP Phones. Advanced, Affordable, Feature-Rich IP Phone for Business or a Home Office
Highlights
- Full-featured 8-line business-class IP phone supporting Power over Ethernet (PoE)
- Connects directly to an Internet telephone service provider or to an IP PBX
- Dual Gigabit Ethernet switched ports, speakerphone, caller ID, call hold, conferencing, and more
- Easy installation and secure remote provisioning, as well as menu-based and web-based configuration
- Supports both SIP and SPCP Protocol with Cisco Unified Communication 500 Series
Rich Phone Features and SIP Compatibility
The Cisco SPA508G 8-Line IP Phone with 2-Port Gigabit Ethernet Switch provides advanced voice and data communications features small businesses need to stay productive and responsive. Based on Session Initiation Protocol (SIP), the Cisco SPA508G has been tested to ensure complete interoperability with leading voice over IP (VoIP) equipment from voice over IP 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.
Stylish and affordable, the Cisco SPA508G IP Phone can also be used with productivity-enhancing features such as VoiceView Express, and Cisco XML Applications when interfacing with Cisco Unified Communication 500 Series in SPCP Mode.
Secure and Easy to Manage
The Cisco SPA508G was designed to simplify installation, while providing enhanced security for complete peace of mind. It utilizes standard encryption protocols to perform secure remote provisioning and unobtrusive in-service software upgrades. Secure 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
- Four voice lines
- Backlit pixel-based display: 128 x 64 monochrome graphical liquid crystal display (LCD)
- Line status: active line indication, 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 (callers hear line busy tone)
- 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 duration and start time stored in call logs
- Call timer
- Name and identity (text) displayed at startup
- Distinctive ringing based on calling and called number
- 12 user-customizable ring tones
- Speed dialing, eight entries
- Configurable dial/numbering plan support
- Intercom*
- Group paging
- NAT Traversal, including STUN support
- DNS SRV and multiple A records for proxy lookup and proxy redundancy
- Debug, syslog, report generation, and event logging
- Secure call encrypted voice communication support
- Built-in web server for administration and configuration with multiple security levels
- Automated remote secure provisioning via TFTP, http or https
- Option to require administrator password to reset unit to factory defaults
* Feature requires support by call server.
Hardware Features
- Pixel-based display: 128 x 64 monochrome LCD graphical display with backlight
- Dedicated illuminated buttons for:
– Audio mute on/off
– Headset on/off
– Speakerphone on/off
- Four-way rocking directional knob for menu navigation
- Voicemail message waiting indicator light
- Voicemail message retrieval button
- Dedicated hold button
- Settings button for access to feature, setup, and configuration menus
- Volume control rocking up/down knob controls handset, headset, speaker, ringer
- Standard 12-button dialing pad
- High-quality handset and cradle
- Built-in high-quality microphone and speaker
- Headset jack: 2.5 mm
- Electronic Hook Switch (EHS) Support with selected Plantronics headsets with adapter
- Gigabit Switch Port and PC Port: 10/100/1000BASE-T RJ-45
- 802.3af compliant Power over Ethernet (PoE)
- Optional 5 VDC universal (100-240V) switching power supply is ordered separately
Regulatory Compliance
- FCC (Part 15, Class B), CE Mark, A-Tick
Security Features
- Password-protected system, preset to factory default
- Password-protected access to administrator and user-level features
- https with factory-installed client certificate
- http digest: encrypted authentication via MD5 (RFC 1321)
- Up to 256-bit Advanced Encryption Standard (AES) encryption
Specifications
- 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 (RFC793)
- UDP - User Datagram Protocol (RFC 768)
- RTP - Real Time Protocol (RFC 1889) (RFC 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 v2 - Session Initiation Protocol version 2 (RFC 3261, 3262, 3263, 3264)
- SPCP - Smart Phone Control Protocol with UC500
- 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 u-law)
- G.726 (16/24/32/40 kbps)
- G.729 A
- G.722
- Dynamic payload support
- Adjustable audio frames per packet
- DTMF - Dual-tone multifrequency, 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
- VAD - Voice activity detection with silence suppression
- Attenuation/gain adjustments
- MWI - Message waiting indicator tones
- VMWI - Voicemail 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
- TR-69, TR-104, and TR-111 Provisioning
- Asynchronous notification of upgrade availability via NOTIFY
- Nonintrusive in-service upgrades
- Report generation and event logging
- Statistics transmitted in BYE message
- Debug and syslog server records: configurable per line
Power supply (Not Included)
- Power supply is optional and is purchased separately
- Models: PA100-NA, PA100-EU, PA100-UK, PA100-AU
- Switching type (100-240V) automatic
- DC input voltage: +5 VDC at 2.0A maximum
- Power adapter: 100-240V50-60 Hz (26-34 VA) AC input
Physical interfaces
- Two 10/100/1000BASE-T RJ-45 Ethernet ports (IEEE 802.3)
- Handset: RJ-9 connector
- Built-in speakerphone and microphone
- Headset 2.5-mm port
Indicator lights/LED
- Speakerphone on/off button with LED
- Headset on/off button with LED
- Mute button with LED
- Message waiting indicator LED
- Voicemail message retrieval button
- Hold button
- LED test function
Dimensions (W x H x D)
0.42 x 8.35 x 1.73 in. (214 x 212 x 44 mm)
Unit weight
2.43 lbs (1.1 kg)
Operating temperature at sea level
32 - 113 F (0 - 40 C)
Storage temperature at sea level
-13 - 185 F (-20 - 70 C)
Operating humidity at sea level
5% to 95% noncondensing
Storage humidity at sea level
5% to 95% noncondensing
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