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PHONE ADAPTER WITH 2 PORTS FOR VOICE-OVER-IP (NORTH AMERICA)

Phone Adapter With 2 Ports For Voice-Over-Ip (North America)
Price: US$73.44
In Stock. Usually ships in 1-2 business days
Platform: Windows
Category: Windows Telephones
By: Cisco Systems - Cobo
Manufacturer Part#: PAP2T-NA
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FEATURES- Internet Phone Adapter with 2 Ports for Voice-over-IP
       Feature-Rich VoIP Service through your High-Speed Internet 
        Connection!
       The Linksys Internet Phone Adapter enables high-quality feature-rich 
        VoIP (voice over IP) service through your broadband Internet 
        connection. Just plug it into your home Router or Gateway and use 
        the two standard telephone ports to connect analog phones or use 
        one of the ports for a fax machine. Each phone port operates 
        independently, with separate phone service and phone numbers, like 
        having two telephone lines. You'll get clear reception and a 
        reliable fax connection, even while using the Internet at the same 
        time. 
       With Internet telephony, along with low domestic and international 
        phone rates, an impressive array of special telephone features are 
        available. Choose your preferred free local dialing area code, 
        regardless of where you live. Or add a virtual telephone number in 
        any area code, forwarded to your Internet phone. You can even add a 
        toll-free number. The Linksys Internet Phone Adapter is compatible 
        with these and all of the other special telephone features that are 
        available from your Internet telephony service provider, such as 
        Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive 
        Ring, and much more. 
       Let the Linksys Internet Phone Adapter enhance your existing 
        Internet connection with a high-quality high-value VoIP service. 
    Includes PAP2T Phone Adapter Unit, Power Adapter, RJ-45 Ethernet Cable,
      Quick Installation Guide.
* Enables feature-rich telephone service over your broadband Internet 
  connection.
* Two standard telephone ports for analog phones or use one of the ports 
  for a fax machine, each with an independent phone number.
* High quality, clear sounding voice service simultaneous with Internet 
  use.
* Two voice ports (RJ-11) for analog phones or Fax machines
* Impedance Agnostics - 8 Configurable Settings
* Call Waiting, Cancel Call Waiting, Call Waiting Caller ID
* Caller ID with Name/Number (Multi-national Variants)
* Caller ID Blocking
* Call Forwarding: No answer, Busy, All
* Do Not Disturb
* Call Transfer
* Three-way Conference Calling with Local Mixing
* Message Waiting Indication - Visual and Tone Based
* Call Return
* Call Back on Busy
* Call Blocking with Toll Restriction
* Delayed Disconnect
* Distinctive Ringing - Calling and Called Number
* Off-hook Warning Tone
* Selective/Anonymous Call Rejection
* Hot line and Warm Line Calling
* Speed Dialing of 8 Numbers/Addresses
* Music on Hold
     
  -- SPECIFICATIONs ------------------------------------
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)
               SNTP - Simple Network Time Protocol (RFC 2030)
VOICE GATEWAY- SIPv2: Session Initiation Protocol v2 (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 
                 (incl. STUN).
               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
               Adjustable Audio Frames per Packet
FAX CAPABILITY- 
               Fax Tone Detection Pass-Through
               Fax Pass-Though - Using G.711
               DTMF: In-band & Out-of-band (RFC 2833) (SIP Info)
               Flexible Dial Plan Support with Interdigit Timers and IP 
                 Dialing.
               Call Progress Tone Generation
               Jitter Buffer - Adaptive
               Frame Loss Concealment
               Full Duplex Audio
               Echo Cancellation (G.165/G.168)
               VAD - Voice Activity Detection with Silence Suppression 
               Attenuation/Gain Adjustments
               Flash Hook Timer
               MWI - Message Waiting Indicator Tones
               VMWI - via FSK
               Polarity Control
               Hook Flash Event Signaling
               Caller ID Generation (Name & Number) - Bellcore, DTMF, ETSI
               Music on Hold Client
               Streaming Audio Server - up to 10 sessions
SECURITY     - Password Protected System Reset to Factory Default
               Password Protected Admin and User Access Authority
               Provisioning/Configuration/Authentication:
                 - HTTPS with Factory Installed Client Certificate
               HTTP Digest - Encrypted Authentication via MD5 (RFC 1321)
               Up to 256-bit AES Encryption
PROVISIONING, ADMINISTRATION & MAINTENANCE-
               Web Browser Administration & Configuration via Integral Web 
                 Server.
               Telephone Key Pad Configuration with Interactive Voice 
                 Prompts.
               Automated Provisioning & Upgrade via HTTPS, HTTP, TFTP
               Asynchronous Notification of Upgrade Availability via NOTIFY
               Non-intrusive, In-Service Upgrades
               Report Generation & Event Logging
               Stats in BYE Message
               Syslog & Debug Server Records - Per Line Configurable
CONNECTORS   - (1) 10BaseT/RJ45 Ethernet port (IEEE 802.3)
               (2) RJ-11 FXS Phone ports (for analog circuit telephone
                     device), (Tip/Ring).
SUBSCRIBER LINE INTERFACE CIRCUIT- (SLIC)
               Ring Voltage       : 40-55 VRMS Configurable 
               Ring Frequency     : 10 Hz - 40 Hz 
               Ring Waveform      : Trapezoidal and Sinusoidal 
               Maximum Ringer Load: 3 REN
               On-hook/off-hook Characteristics:
                  On-hook voltage (tip/ring): -50 V NOMINAL
                  Off-hook current          : 25 mA min
                  Terminating Impedance     : 8 Configurable Settings 
                          including North America 600 ohms, European CTR21.
INDICATORS   - LEDs for Power, Ethernet, Phone1, Phone2
APPROVALS    - FCC (Part 15, Class B), cUL, CE, IC-003, A-Tick
POWER SOURCE - DC Input Voltage: +5 VDC at 2.0 A Max.
               Power Consumption: 5 Watts
               Switching Type (100-240v) Automatic
               Power Adapter: 100-240v - 50-60Hz (26-34VA) AC Input, 1.8m 
                                cord.
DIMENSIONS   - 3.98" x 3.98" x 1.10"            WT.- 5.4 oz.

Linksys Voice Over IP Phone Adapter with 2 Phone Ports.

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