New Unlocked PAP2T SIP VoIP Phone Adapter with 2 FXS Phone Ports VoIP Gateway

Description

Product Name: PAP2T Phone Adapter

Model: PAP2T

Packing List:

1* PAP2T

1* Charger

1* Manual

1* RJ45 Cable

1* Retail box (Optional)

The Link sys Internet Phone Adapter provides 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 for connecting analog phones or using one of the ports for a fax machine. Each phone port operates independently,

with separate phone service and phone numbers, like having two phone 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 call rates, an impressive array of special

Telephone features are available. Choose your preferred free local area code, regardless of where you live.

Or add a virtual phone number with 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 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 version 4 (RFC 791), upgradeable to version 6 (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 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 Gateway
SIPv2: Session Initiation Protocol v2 (RFC 3261, 3262, 3263, 3264)
SIP Proxy Redundancy – Dynamic via DNS SRV and A Records
Re-registration with the Primary SIP Proxy Server
SIP Support in Network Address Translation Networks – NAT (including STUN)
Secure (Encrypted) Calling via a 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
Audio Streaming Server – up to 10 sessions

Provisioning, Administration, and Maintenance:
Web Browser Administration and Configuration via the Integrated Web Server
Telephone Keypad Configuration with Interactive Voice Prompts
Automated Provisioning and Upgrades via HTTPS, HTTP, and 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 – Configurable on a Per-Line Basis