Grandstream HandyTone HT801 ATA

Product Information

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. As mentioned previously, The HT801/HT802 has a built-in voice prompt menu for simple device configuration. Please refer to “ Understanding HT801/HT802 Interactive Voice Prompt Response Menu” for more information about IVR and how to access its menu. Press *** (press the star key three times) to access the IVR menu and wait until you hear “Enter the menu option “.

If you plan to keep the Handy Tone within a private network behind a firewall, we recommend using STUN Server. The following three settings are useful in the STUN Server scenario: If the option “Flash Digit Control” is enabled on web UI, call operation will require different steps as follows:

MATE modem/router

In some cases, a user may want to make phone calls between the phones connected to the ports of the same HT802 when it is used as a standalone unit, without the use of a SIP server. In such cases, users still will be able to make inter-port calls by using the IVR feature.

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. The HT801 support Bellcore style 3-way Conference. To perform the 3-way conference, we assume that the call is established between phone A and B. Phone A (HT801) wants to bring third phone C into conference: Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for dial tone then hang up. Cancel Delayed Call Forward. To cancel Delayed Call Forward, dial “*93”, wait for dial tone, then hang up.The users can now use the IVR system to obtain specific details about the device’s certificate type. Set priority of DTMF methods according to your preference. This setting should be based on your server DTMF setting. Added Local IVR option that announces extension number of the port. [ Understanding HT801/HT802 Interactive Voice Prompt Response Menu] Alternatively, the HT801 can easily be reset to its default settings and configured for connection to another VoIP internet phone service.

Phone A presses Flash (on the analog phone, or Hook Flash for old model phones) to get a dial tone. Dial the number directly and press # (“Use # as dial key” must be configured in web configuration). Once the GXW42xx V2 device is upgraded to 1.0.3.x+, it cannot be downgraded to 1.0.2 x or earlier version.Updated Time Zone option “GMT+01:00 (Paris, Vienna, Warsaw)” to “GMT+01:00 (Paris, Vienna, Warsaw, Brussels). 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. 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.

Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for dial tone then hang up. Grandstream’s HT818 gives users access to the above features and has a total of eight FXS ports. Advantages of Grandstream’s ATA HandyTone Series Phone A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone. 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: A UK number (or transfer of an existing number) is included in the price, but this will only be issued after a direct debit mandate has been set up for continued use of the service after the trial period.The HT801 supports 1 FXS port which can be configured with 1 SIP account, while HT802 supports 2 FXS ports which can be configured with 2 SIP accounts. Please refer to the following steps to register your accounts via web user interface. Note: For UCM63xx, it's recommended to use the latest UCM firmware so it's most compatible with Wave latest versions. If there's more than one VoIP phone/device on the same local network, assigning each different local UDP ports will avoid port conflicts. Note:DP730/DP722/DP720 firmware can be upgraded once the connected DP750/752 base is upgraded to new firmware. The firmware version on DP730/722/720 can be different from the base. Please refer to the release notes for the handset version number. 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.