Linksys Phone Adapter / VoIP GATEWAY / Adatper/ ATA / Linksys PAP2T

Product Description

Product Name: Linksys VOIP ATA

Model: Linksys PAP2T

Linksys PAP2T-2 FXS ports
Support SIP & FAX T. 38
Ethernet Jack-1 XLAN

Best prices, excellent quality
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.

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