In situations where only one analogue phone (or a multi-handset cordless phone base) is to be connected to the VoIP phone service then the lower cost Yeastar TA100 may be a better option. With the HT802, users can access two lines. Moving up in the Series – HandyTone’s HT812 features an integrated gigabit NAT router, which increases speed. This is included at no additional cost and ultimately can lead to a more stable connection. Direct IP calling allows two parties, that is, a FXS Port with an analog phone and another VoIP Device, to talk to each other in an ad hoc fashion without a SIP proxy.

SIP profile through 1 FXS port on HT801, 2 SIP profiles through 2 FXS ports on HT802 and single 10/100Mbps port on both models.Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for dial tone then hang up. The HT801/HT802 supports Bellcore style 3-way Conference. To perform the 3-way conference, we assume that the call is established between phone A and B are in conversation. Phone A(HT801/HT802) wants to bring third phone C into conference:

The following guide will be a useful guideline to help configure Grandstream adapters with your sipgate account and applies to all adapters from the HT series, e.g. HT 801 / 802 / 812 / 814 etc. Alternatively, the HT801 can easily be reset to its default settings and configured for connection to another VoIP internet phone service.Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729), *027201 (iLBC). *02722 (G722). Insert the ethernet cable into the internet or LAN port of the HT801 and connect the other end of the ethernet cable to an uplink port (a router or a modem, etc.)

From our assessment of the ATA, setup is easy, as is management. For businesses looking to maximize productivity and efficiency of overall communication – the HandyTone Series will more than suit your interests. The Power LED lights up when the HT801/HT802 is powered on and it flashes when the HT801/HT802 is booting up. When placing the analog phone that is connected to the FXS port off hook, the dial tone will be played even if the sip account is not registered. If users prefer the busy tone to be played instead, the following configuration should be made:Upgrade protocol for firmware and configuration update. Press “ 9” to toggle between TFTP / HTTP / HTTPS / FTP / FTPS. Default is HTTPS. You'll find the correct Firmware Server Path for your Grandstream at: www.grandstream.com/support/firmware Testing your Phone's setup: Follow the instructions in your device manual to access your Grandstream Adapter's Web Interface menus.