Linksys PAP2T voip gateway

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
• Syslog & Debug Server Records-Per Line Configurable

Refurbished