1. Linksys PAP2T-2 FXS ports
2. Support SIP & FAX T.38
3. Ethernet Jack-1 XLAN
4. 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 .

 

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

S og & Debug Server Records-Per Line Configurable