Grandstream HandyTone HT801 ATA

Product Information

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. Updated Time Zone option “GMT+01:00 (Paris, Vienna, Warsaw)” to “GMT+01:00 (Paris, Vienna, Warsaw, Brussels). The HT801 is supplied with a UK mains power supply and adaptors which enable UK phones to be connected. If phone C answers the call, phones A and C are in conversation. Then A can press “Flash + 4” to complete transfer. This section documents significant changes from previous versions of the user guide for HT801/HT802. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here.

Cancel Unconditional Call Forward. To cancel “Unconditional Call Forward”, dial “*73”, wait for dial tone, then hang up. Assuming that phones A and B are in conversation. The phone A wants to blind transfer phone B to phone C: Note: Please use the 1.0.9.x firmwares below for GWN7600/7600LR/7610with GWN7000 1.0.9.4deployment. If the GXP21xx phone is on firmware 1.0.7.x, please upgrade to 1.0.8.x first, and then upgrade to 1.0.9.x.

MATE modem/router

Although the Grandstream HT801 adaptor can be connected to any VoIP phone service, we supply it pre-configured for connection to the Premitel VoIP Phone Service. The computer must be connected to the same sub-network as the HT801. This can be easily done by connecting the computer to the same hub or switch as the HT801. CONFIGURING HT801 THROUGH VOICE PROMPTS

Set priority of DTMF methods according to your preference. This setting should be based on your server DTMF setting. For GXW42xx V2 device on 1.0.3.1 or higher version to upgrade to 1.0.17.1, the HTTP firmware upgrade path is: The screenshots below are taken from an older firmware version. With newer version adaptors, the field entries are still correct information entries it's just that some fields have moved and are now in the FXS port tab. With the later firmware editions, please make sure to set "enable port" to YES before saving. When all the FXS ports are registered (for HT802), the simultaneous ring will have one second delay between each ring on each phone. Rebooting the HT80x from RemoteThe HT801 is designed to connect to a VoIP phone service where a traditional landline is not required or is not available (eg fibre-to-the-premises broadband connections).

Please refer to the user manual for more information about IVR and how to access its menu options. REBOOTING HT801 Using Web UI There are 3 LED buttons on HT801 and 4 LED buttons on HT802 that help you manage the status of your Handy Tone. During an active call, press “Flash” button on the analog phone (if the phone has that button) to place the call on hold. The Ethernet LED lights up when the HT801/HT802 is connected to your network through the Ethernet port and it flashes when there is data being sent or received. Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for dial tone then hang up.The HT801 is designed for easy configuration and easy installation, to connect your HT801, please follow the steps below: The users can now use the IVR system to obtain specific details about the device’s certificate type. Dial an extension directly on the same proxy, (e.g. 1008), and then press the # or wait for 4 seconds; The Grandstream HT801 is ideal for situations in which multiple analogue phones (or a loud bell) need to be connected to a VoIP phone service (see description below for more information).

GXW42xx V2 has "V2" printed for model name. For example, "Model: GXW4216 V2". If the model name doesn't have V1 or V2 printed, the device is GXW42xx V1.

That’s a wrap

Note: For UCM63xx, it's recommended to use the latest UCM firmware so it's most compatible with Wave latest versions. Although the HT801 adaptor can be connected to any VoIP phone service, we supply it pre-configured for connection to the Premitel VoIP Phone Service. SIP User ID: User account information, provided by VoIP service provider (ITSP). Usually in the form of digit like phone number or a phone number.