Support NAT traversal using IETF STUN and symmetric RTP (compatible with Cisco's ATA-186, etc)
Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
Provides 1 LAN port and 1 FXS interface for any analog telephones, cordless phones, and fax machines
Support transparent Fax pass-through and in the future T.38 (pending)
Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
Advanced and patent pending adaptive jitter buffer control, packet delay and loss concealment technology
Support popular codecs including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, and G.728. Dynamic negotiation of codec and voice payload length
Support standard voice features such as Caller ID Display or Block, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, early dial, click-to-dial
Support acoustic echo cancellation, voice mail with indicator, downloadable ring tone (pending)
Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
Support DIGEST authenication and encryption using MD5 and MD5-sess.
Provide easy configuration thru manual operation (attached analog phone keypad and voice prompt, Web interface) or personalized automated provisioning via central configuration file for mass deployment.
Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs.
Support for fail-over SIP server and DNS server (pending)
Getting Started Whats included: Telephone Adapter Ethernet Cable AC Power Adapter Equipment you will need: Broadband Connection: DSL, cable modem, or other high-speed Internet connection Router (optional): To share your broadband connection with more than one device. A router is not necessary if you only use one computer. Traditional Analog Telephone To use enhanced features, you will need: Computer Web Browser: MS Internet Explorer (ver. 4.x or higher) or Netscape Navigator (ver. 4.x or higher) About the Telephone Adapter :The Telephone Adapter enables you to connect your phone and either one or more computers to your broadband network. When connecting to a single computer, you do not need a router.
BEFORE PROCEEDING: If you are using DSL that requires PPPoE authentication, please follow these important steps.
Installing the Telephone Adapter on a SINGLE Computer
1.Power off your DSL or cable modem and computer 2.Insert the Ethernet cable into the WAN port. Connect the other end of the Ethernet cable into your DSL or cable modem. 3.Insert an end of another Ethernet cable into the LAN port. Connect the other end of the Ethernet cable into the Ethernet port on your computer. 4.Insert the line from your telephone handset into the PHONE port of the Telephone Adapter. Connect the other end of the telephone cable to a traditional analog telephone. 5.Apply power to DSL or cable modem. Ensure proper status indicators are lit. 6.Plug the power adapter into the Telephone Adapter. The LED on top of the adapter will blink for up to one minute. 7.Apply power to your computer. 8.You can now make calls! Pick up the phone receiver and listen for a dial tone.
The HandyTone 503 offers the next generation of powerful, affordable, high quality and manageable IP telephony ATA/IAD. The integration of a FXO and FXS port enables remote call origination and termination to and from the PSTN line, known as "hop-on and hop-off" calling. This functionality coupled with its compact size makes it an ideal solution for road warriors seeking the savings and benefits of voice-over-ip communication.
» Based on SIP standard and interoperable with most 3rd party SIP compliant devices and software
» Dual 10M/100Mbps network ports, port status and message waiting LED, and a base stand for vertical positioning
» Traditional and advanced telephony features including mutli language voice prompts and transfer to or forward to IP or PSTN
The HT503 has an FXS and an FXO so that it sits between our phones and our land-line. This means I can pick up the phone and place a call on the public switched telephone network (PSTN) or on the Internet depending on how I dial. It also means that calls coming in from the Internet or the PSTN all ring the same phones. It can also do fun things like let me call in from the Internet and place a local call on my land-line but I don’t use those features.
Actually there was one more brief problem where an incoming PSTN call would ring but when the person answering the call picked up they would hear a busy tone and the caller would continue to hear ringing.When the HT503 makes a connection between it’s FXO and it’s FXS it does so by having the one port call the other on the local loopback interface. One thing it means is that if you select a lossy codec for both interfaces it seems to actually encode and decode the audio even though it’s on the same device.
Features & Benefits
Features
1 FXS telephone port (RJ11), 1 FXO PSTN line port (RJ11) with power-outage life line support,
Up to 2 SIP account profiles, SIP over TCP/TLS, SRTP
Dual 10/100 Mbps network ports (RJ45) with integrated high performance NAT router
Status LED for power, telephone, PSTN line, network, and message waiting indication
Advanced telephony features
caller ID from both IP and PSTN,
call waiting, 3-way conference with IP and/or PSTN,
remote call origination and termination from/to PSTN,
hop-on and hop-off calling,
transfer to OR forward to IP or PSTN,
do not disturb,
message waiting indication,
multi-language voice prompts,
flexible dial plan, direct IP calling
Comprehensive voice codecs
Secure and automated provisioning using HTTP/HTTPS/Telnet/TFTP
Symmetric and asymmetric voice codec/RTP in any call sessions
T.38 Fax
SIP over TCP/TLS
Benefits
IP connectivity for any phone and fax
Hop-on/Hop-off calling
Web management for easy configuration and installation
Offers traditional and advanced telephony features
Portable and compact for use at home or on the road
Feature Specifications
Ethernet Ports
2 RJ45 (LAN/WAN)
NAT/Router
Yes
DHCP
Client/Server
FXS Port
1
FXO Port
1
PSTN Pass-through Port
Yes
Voice Mail Indicator
Yes
Voice Codec
G.711(a/u-law), G.723.1, G.729A/B/E, G.726-40/32/24/16 and iLBC, T.38 fax
Remote Configuration
HTTP/HTTPS/Telnet/TFTP Provisioning
Technical Specifications
Telephone Interfaces
1 FXS telephone port (RJ11), 1 FXO PSTN line port (RJ11) with lifeline support
Network Interfaces
Two (2) 10M/100 Mbps ports (RJ-45) with integrated NAT router
LED Indicators
Power, WAN, LAN, PHONE and LINE
Reset Button
Factory reset button
Telephony Features
caller ID display or block, call waiting wither caller ID, flash, blind or attended call transfer, hold, call forward, do not disturb, 3-way conferencing
Voice Codec
G.711 Annex I (PLC) and Annex II (VAD/CNG format), G.723.1A, G.729A/B/E, G.726-40/32/24/16, iLBC
Voice over Packet
Capabilities
Voice Activity Detection (VAD) with Comfort Noise Generation (CNG) and Packet Loss Concealment (PLC), Dynamic Jitter Buffer, G.168 compliant Line Echo Cancellation
Fax over IP
T.38 compliant Group 3 Fax Relay up to 14.4kpbs, Fax Datapump V.17, V.19, V.27ter, V.29 for T.38 fax relay
The HT502 is a powerful VoIP router. The product's inclusion of an integrated high performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices. In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and advanced telephony features.
» Enhanced security
» Automated provisioning using symmetric and asymmetric voice
» Support for a broad range of popular voice codec
Features & Benefits
Universal Plug-in-Play (UPnP)
2 FXS ports (RJ11) w/up to 2 SIP account profiles
Dual10/100 Mpbs ports (RJ45) w/integrated router
Advanced features:
-caller ID, call waiting, 3-way conference, blind or attended transfer
-call forward, do not disturb, voicemail, MLS voice prompts
-T.38 fax, flexible dial plan, direct IP calling
Supports Voice Codecs:
-G.711(a/u-law), G.723.1, G.729A/B, G.729E
-G.726-40/32/24/16 and iLBC
T.38 Fax
HTTP/HTTPS(pending)/Telnet/TFTP Provisioning
SIP over TCP/TLS
IP connectivity for any phone and fax
Web management for easy configuration and installation
Offers traditional and advanced telephony features
Portable and compact for use at home or on the road
Voice Activity Detection (VAD) with CNG (comfort noise generation) and PLC (packet loss concealment), Dynamic Jitter Buffer, Modem detection & auto-switch to G.711,
Packetized Voice Protocol Unit (supports RTP/RTCP and AAL2 protocol), G.168 compliant Echo Cancellation, LEC (line echo cancellation) with NLP
Voice Compression
G.711 + Annex I (PLC), Annex II (VAD/CNG format) encoder and decoder, G.723.1A, G.726(ADPCM), G.729A/B/E, iLBC G.726 provides proprietary VAD, CNG, and signal power estimation Voice Play Out unit (reordering, fixed and adaptive jitter buffer, clock synchronization), AGC (automatic gain control), Status output, Decoder controlling via voice packet header
DHCP Server/Client
Yes, NAT Router or Switched Mode
Fax over IP
T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-through (pending), Fax Datapump V.17, V.19, V.27ter, V.29 for T.38 fax relay
QoS
Diffserve, TOS, 802.1 P/Q VLAN tagging
IP Transport
RTP/RTCP
DTMF Method
Flexible DTMF transmission method, user interface of In-audio, RFC2833, and/or SIP Info
IP Signaling
SIP (RFC 3261)
Provisioning
TFTP, HTTP, HTTPS (pending)
Control
TLS/SIPS
Management
Syslog support, HTTPS and telnet (pending), remote management using Web browser, Auto/manual provisioning system Support Layer 2 (802.1Q, VLAN, 802.1p), Layer 3 QoS (Tos, DiffSery, MPLS)
Power
Output: 12VDC, 0.5A / Input: 100-240 VAC/50-60 Hz
Environmental
Operational: 32°-104°F or 0°-40°C
Storage: 10°-130°F / Humidity: 10-90% Non-condensing
Dimensions
(H x W x D)
25mm x 115mm x 75mm (when laying flat); 115mm x 25mm x 75mm (standing up)
Short & long haul
REN3: Up to150 ft on 24 AWG line
Call Handling Features
Caller ID display or block, Call waiting caller ID, Call waiting/flash, Call transfer, hold, forward, mute, 3-way conferencing
Caller ID
Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID
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