-20%
Linksys PAP2T voip gatewayLinksys PAP2T voip gateway
Linksys PAP2T voip gateway 2Linksys PAP2T voip gateway 2

Linksys PAP2T voip gateway

49.95 39.95 incl. 21% BTW

The Linksys Internet Phone Adapter enables high-quality feature-rich VoIP service through your broadband Internet connection . Just Plug it into your home or office 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 avialable . 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 tollferr number . The Linksys Internet Phone Adapter is compatible with these and 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 .

Data Networking

  • MAC Address (IEEE 802.3)
  • IPv4-Internet Protocol v4 (RFC 791 ) upgrateale to v6 (RFC 1883)
  • ARP-AddressResolution 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 (RFC 793)
  • UDP-User Datagram Protocol (RFC 768)
  • RTP-Real Time Protocol (RFC 1889)
  • Diffserv (RFC 2475) , Type of Service-TOS (RFC 791/1394)
  • 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 u-law)
  • G.726 (16 / 24 /32 /40 kbps)
  • G.729A
  • G.723.1 (6.3 kbps , 5.3 kbps)
  • Dynamic Payload
  • Adjustment Audio Frames per Packet

Fax Capability

  • Fax Tone Detection and Pass-through (Using .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 Silence Suppression
  • Flash Hook Timer
  • MWI-Message Waiting Indicator Tone
  • VMWI-Visual Message Waiting Indicator 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

Provisioning , Administration & Maintenance :

  • Web Browser Administration & Configuration via Integrated Web Server
  • Telephone Key Pad Configuration with Interactive Voice Prompts
  • Automted Provisioning & Update Availability via SIP Notify
  • Non-intrusive , In-Service Upgrades
  • Report Generation & Event Logging
  • Stats in BYE Message
  • Syslog & Debug Server Records-Per Line Configurable

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Extra informatie

Gewicht 1 kg

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