LINKSYS PAP2T / VOIP ATA / LINKSYS phone adapter

Product Name: Linksys Phone Adapter Mode: PAP2TThe Linksys Internet Phone Adapter enables 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 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 available. 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 toll-free number. The Linksys Internet Phone Adapter is compatible with these and all 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) 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 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 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 PacketFax 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