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Telefonia / Voip - Gateways

SPA922-NA Telefone VoIP Linksys com 2 Portas Lan

1 Linha e Visor LCD (Indisponível - sem previsão)

SPA922-NA Telefone VoIP Linksys com 2 Portas Lan

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envio de 30 a 60 dias uteis

R$ 784,00

à vista no Pix ou Boleto

ou R$ 824,91 em 1x no cartão (via site)

ou R$ 870,74 no cartão em até 10x sem juros (apenas na loja física)

R$ 784,00

à vista no Pix ou Boleto

R$ 824,91
em 1x no cartão (via site)

R$ 870,74 no cartão
em até 10x sem juros
(apenas na loja física)

Preço e prazo desatualizados, solicite revisão antes de comprar

Comprar

Preço e prazo desatualizados, solicite revisão antes de comprar

Condição:

Novo em embalagem comercial

Garantia:

12 meses Assist. Técnica do Fabricante

NCM:

85176254

R$ 784,00

à vista no pix

Comprar

Condição:
Novo em embalagem comercial

Garantia:
12 meses Assist. Técnica do Fabricante

NCM: 85176254

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SPA922-NA Telefone VoIP Linksys com 2 Portas Lan
1 Linha e Visor LCD (Indisponível - sem previsão)

Condição de SPA922-NA: Novo em embalagem comercial

Garantia de SPA922-NA: 12 meses Assist. Técnica do Fabricante

Stylish and functional in design, the Linksys SPA-922 Single Line SIP Phone Dual RJ45, PoE IP phone is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large-scale IP Centrex deployment.

The SPA922 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP phone that is unparalleled in features, value, and support.

Detailed Description

Standard features on the SPA-922 include dual switched Ethernet ports, 802.3af PoE, a high resolution graphical display, speakerphone and a 2.5 mm head-set port. The SPA922 supports one line with two call appearances and provides support for three way conferencing, attended call transfer, and placing a call on hold to answer an incoming call. The line can be configured as a unique phone number (or extension), or can be configured to share a number that is assigned to multiple phones.

Full featured one-line business class IP phone supporting Power over Ethernet 802.3af

Connect directly to an Internet Telephone Service Provider or connect to an IP PBX

Dual switched Ethernet ports, Speakerphone, Caller ID, Call Hold, Conferencing, and more

Easy installation with secure remote provisioning. Menu based and web based configuration.

Comprehensive Interoperability and SIP Based Feature Set
Based on the SIP standard, the SPA-922 has been tested to ensure comprehensive interoperability with equipment enabling service providers to quickly roll-out competitive, feature rich services to their customers. With hundreds of features and configurable sevice parameters, the SPA922 addresses the requirements of traditional business users while leveraging the advantages of IP telephony. Features such as easy station moves, presence, and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA-922. Carrier-Grade Security, Provisioning, and Management
The SPA922 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Linksys 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 hassle and expense of managing, preloading, and re-configuring customer premise equipment (CPE).

Package Contents

1 - SPA-922 IP Phone, Handset, and Stand
1 - Handset Cord
1 - RJ45 Ethernet Cable
1 - Quick Installation Guide
Power Supply is Included

Telephony Features

One Voice Line with Two Call Appearances
Backlit Pixel Based Display: 128x64 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 and Outbound Caller ID Blocking
Outbound Caller ID Blocking
Call Transfer - Attended and Blind
Three Way Call Conferencing with Local Mixing
Connects to External Conference Bridge for Multi-party
Conferencing
Automatic Redial of Last Calling and Last Called Numbers
On-Hook Dialing
Call Pick Up - 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
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
Call Number using Name - Directory Matching or via Caller ID
Subsequent Incoming Calls with Calling Name and Number
Date and Time with Intelligent Daylight Savings Support
Call Duration and Start Time Stored in Call Logs
Call Timer
Name and Identity (Text) Displayed at Start Up
Distinctive Ringing Based on Calling and Called Number
Ten User Downloadable Ring Tones - Ring Tone Generator Free from www.linksys.com
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
Syslog, Debug, 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 Provisioning, Multiple Methods. Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP)
Optionally Require Admin Password to Reset Unit to Factory Defaults
** Feature requires support by call server

Hardware Features

Pixel Based Display: 128x64 Monochrome LCD Graphical Display with back light
Dedicated Illuminated Buttons for:
Audio Mute On/Off
Headset On/Off
Speakerphone On/Off
Four Way Rocking Directional Knob for Menu Navigation
Voice Mail Message Waiting Indicator Light
Voice Mail Message Retrieval Button
Dedicated Hold Button
Settings Button for Access to Feature, Set-up, 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 millimeter
LED Test Function
Two Ethernet LAN Ports with Integrated Ethernet Switch 100BaseT RJ-45
802.3af Compliant Power over Ethernet (PoE)

Data Networking

MAC Address (IEEE 802.3)
IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 1883)
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 (RFC792)
TCP - Transmission Control Protocol (RFC793)
UDP - User Datagram Protocol (RFC768)
RTP - Real Time Protocol (RFC 1889) (RFC 1890)
RTCP - Real Time Control Protocol (RFC 1889)
DiffServ (RFC 2475), Type of Service - TOS (RFC 791/1349)
VLAN Tagging 802.1p/q - Layer 2 QoS
SNTP - Simple Network Time Protocol (RFC 2030)

Voice Gateway

SIPv2 - Session Initiation Protocol Version 2 (RFC 3261, 3262, 3263, 3264)
SIP Proxy Redundancy - Dynamic via DNS SRV, A Records
Re-registration with Primary SIP Proxy Server
SIP Support in Network Address Translation Networks - NAT (including STUN)
SIPFrag (RFC 3420)
Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP
Codec Name Assignment
Voice Algorithms:
- G.711 (A-law and ?-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
DTMF: In-band and Out-of-Band (RFC 2833) (SIP INFO)
Flexible Dial Plan Support with Inter-Digit 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 - Voice Mail Waiting Indicator - Via NOTIFY, SUBSCRIBE
Caller ID Support (Name and Number)
Third Party Call Control (RFC 3725)

Provisioning, Administration & Maintenance

Integrated Web Server Provides Web Based Administration and Configuration
Telephone Key Pad Configuration via Display Menu / Navigation
Automated Provisioning and Upgrade via HTTPS, HTTP, TFTP
Asynchronous Notification of Upgrade Availability via NOTIFY
Non-intrusive, In-Service Upgrades
Report Generation and Event Logging
Statistics Transmitted in BYE Message
Syslog and Debug Server Records - Configurable Per Line

Physical Interfaces

2 100baseT RJ-45 Ethernet Port (IEEE 802.3)
Handset: RJ-7 Connector
Built-in Speakerphone and Microphone
Headset 2.5 mm Port

Tags de SPA922-NA: SPA922-NA, SPA922, Linksys

SPA922-NA Telefone VoIP Linksys com 2 Portas Lan 1 Linha e Visor LCD (Indisponível - sem previsão)

Condição de SPA922-NA:
Novo em embalagem comercial

Garantia de SPA922-NA:
12 meses Assist. Técnica do Fabricante

Stylish and al in design, the Linksys SPA-922 Single Line SIP Phone Dual RJ45, PoE IP phone is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large-scale IP Centrex deployment.

The SPA922 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP phone that is unparalleled in features, value, and support.

Detailed Description

Full featured one-line business class IP phone supporting Power over Ethernet 802.3af

Connect directly to an Internet Telephone Service Provider or connect to an IP PBX

Dual switched Ethernet ports, Speakerphone, Caller ID, Call Hold, Conferencing, and more

Easy installation with secure remote provisioning. Menu based and web based configuration.

Comprehensive Interoperability and SIP Based Feature Set
Based on the SIP standard, the SPA-922 has been tested to ensure comprehensive interoperability with equipment enabling service providers to quickly roll-out competitive, feature rich services to their customers. With hundreds of features and configurable sevice parameters, the SPA922 addresses the requirements of traditional business users leveraging the advantages of IP telephony. Features such as easy station moves, presence, and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA-922. Carrier-Grade Security, Provisioning, and Management
The SPA922 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Linksys 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 hassle and expense of managing, preloading, and re-configuring customer premise equipment (CPE).

Package Contents

1 - SPA-922 IP Phone, Handset, and Stand
1 - Handset Cord
1 - RJ45 Ethernet Cable
1 - Quick Installation Guide
Power Supply is Included

Telephony Features

One Voice Line with Two Call Appearances
Backlit Pixel Based Display: 128x64 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 and Outbound Caller ID Blocking
Outbound Caller ID Blocking
Call Transfer - Attended and Blind
Three Way Call Conferencing with Local Mixing
Connects to External Conference Bridge for Multi-party
Conferencing
Automatic Redial of Last Calling and Last Called Numbers
On-Hook Dialing
Call Pick Up - 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
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
Call Number using Name - Directory Matching or via Caller ID
Subsequent Incoming Calls with Calling Name and Number
Date and Time with Intelligent Daylight Savings Support
Call Duration and Start Time Stored in Call Logs
Call Timer
Name and Identity (Text) Displayed at Start Up
Distinctive Ringing Based on Calling and Called Number
Ten User Downloadable Ring Tones - Ring Tone Generator Free from www.linksys.com
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
Syslog, Debug, 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 Provisioning, Multiple Methods. Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP)
Optionally Require Admin Password to Reset Unit to Factory Defaults
** Feature requires support by call server

Hardware Features

Pixel Based Display: 128x64 Monochrome LCD Graphical Display with back light
Dedicated Illuminated Buttons for:
Audio Mute On/Off
Headset On/Off
Speakerphone On/Off
Four Way Rocking Directional Knob for Menu Navigation
Voice Mail Message Waiting Indicator Light
Voice Mail Message Retrieval Button
Dedicated Hold Button
Settings Button for Access to Feature, Set-up, 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 millimeter
LED Test Function
Two Ethernet LAN Ports with Integrated Ethernet Switch 100BaseT RJ-45
802.3af Compliant Power over Ethernet (PoE)

Data Networking

MAC Address (IEEE 802.3)
IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 1883)
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 (RFC792)
TCP - Transmission Control Protocol (RFC793)
UDP - User Datagram Protocol (RFC768)
RTP - Real Time Protocol (RFC 1889) (RFC 1890)
RTCP - Real Time Control Protocol (RFC 1889)
DiffServ (RFC 2475), Type of Service - TOS (RFC 791/1349)
VLAN Tagging 802.1p/q - Layer 2 QoS
SNTP - Simple Network Time Protocol (RFC 2030)

Voice Gateway

SIPv2 - Session Initiation Protocol Version 2 (RFC 3261, 3262, 3263, 3264)
SIP Proxy Redundancy - Dynamic via DNS SRV, A Records
Re-registration with Primary SIP Proxy Server
SIP Support in Network Address Translation Networks - NAT (including STUN)
SIPFrag (RFC 3420)
Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP
Codec Name Assignment
Voice Algorithms:
- G.711 (A-law and --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
DTMF: In-band and Out-of-Band (RFC 2833) (SIP INFO)
Flexible Dial Plan Support with Inter-Digit 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 - Voice Mail Waiting Indicator - Via NOTIFY, SUBSCRIBE
Caller ID Support (Name and Number)
Third Party Call Control (RFC 3725)