Zoom 5801 Voice over IP Telephone Adapter 58010011 Merkblatt

The economical Zoom Model 5801 and Model 5802 Voice over IP Telephone Adapters allow standard analog
telephones  to make and receive calls over a broadband Internet connection. Simply plug a conventional or cordless
phone into the adapter and then plug the telephone adapter into a broadband-connected router or Internet gateway.
Up to five and often more attached phones can be rung simultaneously using the built-in ring generator.

The Model 5801 and Model 5802 support industry-standard Session Initiation Protocol (SIPv2) and work with a wide
range of service providers and SIP-based VoIP equipment.

Both models provide one FXS and one FXO port and allow calls to be bridged between the Public Switched Telephone
Network (PSTN) and Voice Over IP.

The FXO port of the adapters also include the TelePort

™

, an intelligent relay that allows a single phone to place and

receive both VoIP calls and calls over the PSTN. PSTN fail-over allows the 5801 and 5802 to automatically route calls
over the dial-up phone network when power is lost. PSTN support also allows emergency dialing using services like
911, 112, or 999 calling. PSTN to VoIP call bridging with security is provided, as well as VoIP to PSTN call bridging.

Voice over IP Service providers can deliver a rich variety of advanced telephone services through the Model 5801 or
5802 including CLASS features such as Call Waiting, Caller Identification, Call Transfer, Call Hold, Call Forwarding,
Distinctive Ringing, and Voice Message Waiting Indication.

The VoIP telephone adapters can be configured remotely using a TFTP or HTTP download from the service provider and
updates to the firmware can be automatically delivered. Local configuration is done with a browser-based graphical
user interface.

The Model 5801 supports G.711, iLBC and G.729 voice codecs. The Model 5802 supports G.711 and iLBC.

Both units are built and supported by Zoom Technologies, a publicly traded company (NASDAQ:ZOOM) with over 29
years of experience in telephony and data communications.

Analog Telephone Ports

• One FXS type Loop-start interface with RJ-11
• One FXO analog interface with Teleport and RJ-11
• Up to 5 REN (Ringer Equivalence Number), will ring up to 5 typical telephones

and often more

• Programmable Ring Patterns
• Call progress tones supported: Initial dial tone, Secondary dial tone, Stuttered dial tone,

Message waiting dial tone, Call forward dial tone, Pre-ringback dial tone, Ring back tone,
Call waiting tone, Call holding tone, Call disconnect tone, Call conference tone,
Busy tone, Reorder tone (network busy/fast busy), Off hook warning.

• Power Fail Over
• Auto switch to PSTN for emergency calling using 911 and other programmed three digit

codes

• VoIP to PSTN call bridging, PSTN to VoIP call bridging with ANI security

Status Indicators

•

Power, LAN link, VoIP ready, VoIP call-in-progress, PSTN port ready,

Voice message waiting

Voice over IP (VoIP)

•

SIPv2 - Session Initiation Protocol (RFC 3261, 3262, 3263, 3264)

Protocols

•

SDP - Session Description Protocol (RFC 2327)

•

RTP - Real Time Protocol (RFC 1889, 1890)

• RTCP - Real-Time Control Protocol (RFC 1889)
• X-NSE - Tone Events for SIP/RTP (RFC 2833)
• AVT - Tone Events for SIP/RTP (RFC 2833)
• Power-on Auto Registration
• Re-registration with SIP Proxy Server
• SIP over UDP
• SIP authentication (HHP Digest with MD5)

Model 5801 

with G.711, iLBC and G.729 Codecs

Model 5802 

with G.711 and iLBC Codecs