Grandstream HandyTone 286

Grandstream HandyTone ATA-286 is an award-winning next generation VoIP analog telephone adaptor based on industry open standards. Built upon innovative technology, Grandstream HandyTone ATA-286 features market leading superb sound quality, compact size, and rich functionalities at highly-affordable price.

Grandstream HandyTone-286 Single FXS Port Analog Adapter for VoIP Phone Features:

• Support SIP 2.0, TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols
• 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 authentication 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)

HandyTone Series comparison table

* FXO port allows local and remote PSTN-to-VoIP call origination and VoIP-to-PSTN call termination, automated call routing through PSTN, and power fail-over. PSTN Pass-through allows local PSTN line service for income/outbound call and failover. Click here for additional information