VOIP Phone Adapter VoIP phone Linksys(35$)
PAP2T LINKSYS PAP2T-NA SIP VOIP Phone Adapter VoIP phone Linksys PAP2T Internet Phone Adapter Two Phone Ports without retail box.
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Linksys PAP2T, Voip Sip Adapter PAP2T
Procuts Name: VoIP Phone Adapter
Product Model: Linksys PAP2T
Package details :
1* Linksys PAP2T (no retail box, but well pack, no worries, if need that box,pls add 2usd)
1*Chargers
1* Manual
1*RJ-45 usb cable
1. 2 FXS VoIP phone adapter , 1 RJ-45 port LAN Ethernet
2. SIPv2: Session Initiation Protocol v2 (RFC 3261, 3262, 3263, 3264)
3. Fax Tone Detection Pass-Through
4. Call Waiting,Transfer,Call Forwarding: No Answer / Busy / All
5. Call Return,Call Back on Busy,Call Blocking with Toll Restriction
6. Supports G.711, G.726, G.729, and G.723.1 codec
7. Supports DTMF tone detection and generation
8. Web-based configuration through a built-in web server
Model : PAP2T
* Note: Many specifications are programmable within a defined range or list of options.
Please see the PAP2T Administration Guide for details. The configuration profile is uploaded
to the SPA3000 at the time of provisioning.
Data Networking
MA- 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 791/1394)
SNTP- Simple Network time protocol (RFC 2030)
Voice
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 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 Adjustments
Flash Hook Timer
MWI – Message Waiting Indicator Tones
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
Automated Provisioning & Upgrade via HTTPS, HTTP, TFTP
Asynchronous Notification of Upgrade Availability via SIP NOTIFY
Non-intrusive, In-Service Upgrades
Report Generation & Event Logging
Stats in BYE Message
Syslog & Debug Server Records – Per Line Configurable
Physical Interfaces:
1 10baseT RJ-45 Ethernet Port (IEEE 802.3)
2 RJ-11 FXS Phone Ports – For Analog Circuit Telephone Device (Tip/Ring)
Indicator Lights/LED:
Phone1, Phone2, Internet, Power
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