Linksys Internet Phone Adapter

Product Code: Linksys
Availability: In Stock
Weight: 1.00 kg
Size (LxWxH): 10.10 cm x 3.10 cm x 10.10 cm
Condition: New
Price: RM149
Qty:  

Features:

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 the Linksys Internet Phone Adapter,along with low domestic and international phone rates,a n 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.

Product Characteristics:

  • Support SIP 2.0 (RFC3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP,DNS, DHCP, NTP, PPPoE, STUN, and TFTP etc
  • Inside Rooter/NAT, gateway, and DMZ Port transmit
  • Support the PETN making or receiving calls (by the FXO Port)
  • Use the DSP CNOS chip to keeping a wonderful audio-quality, advanced shaking control, and the technology of hiding the lost message
  • Support kinds of voice coding, including G.711 (alaw and u-law), G.723.1 (5.3k/6.3k), G.726 (40K/ 32K/ 24K/ 16K), G.29 A/B, and iLBC
  • Support incoming call on show, restricting, and holding, disconnection, call transfer, call divert, DTMF, dialing project, etc
  • Support conference call
  • Support passing through
  • Support layer 2 (802.1Q VLAN, 802.1p) and layer 3(QoS,DiffServ, Tos)
  • Support NAT auto-penetrating, no need to modify the setting of the NAT
  • Support configuration files by inside IVR equipment, Web browser, or TFTP and HTTP Center Server
  • Support upgrading encrypts configuration files by TFTP or HTTP
  • Microminiaturize and legerity design (size as a wallet), a adapter that convenient for schlepping
     

Specifications:

  • Two voice ports(RJ-11) for analog phones or FAX machine
  • Impedance Agnostics-8 Configurable Settings
  • Call Waiting, Cancel Call Waiting, Call Waiting Caller ID
  • Caller ID with Name/ Number (Multi-national Variants)
  • Caller ID Booking
  • Call Forwarding: No answer, Busy, All
  • Do not disturb
  • Call Transfer
  • 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
  • Hotline and Warm Line Calling
  • Speed Dialling of 8 Numbers/ Addresses
  • Music on hold

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 (RFC 792)
  • TCP- Transmission Control Protocol (RFC 793)
  • UDP- User Datagram Protocol (RFC 768)
  • RTP -Real Time Protocol (RFC 18890 ( 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 μ-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 Packer​

Fax Capability:

  • Fax Tone Detection Pass-Through
  • Fax Pass-Through – Using G.711
  • DTMP: In-band & Out of band (RFC 2833) (SIP info)
  • Flexible Dial Plan Support with Interdigit timers and IP Dialling
  • 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

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