Protecting your private information is our priority. This Statement of Privacy applies to https://twistedvoip.com and Twisted VoIP LLC and governs data collection and usage. Unless otherwise noted, all references to Twisted VoIP for this Privacy Policy include https://twistedvoip.com and Twisted VoIP LLC. The Twisted VoIP website is an eCommerce site. Using the Twisted VoIP website, you consent to the data practices described in this statement.
Collection of your Personal Information
To better provide you with products and services offered, Twisted VoIP may collect personally identifiable information, such as your:
First and Last Name
Mailing Address
E-mail Address
Phone Number
Employer
Job Title
Physical Address
If you purchase Twisted VoIP’s products and services, we collect billing and credit card information. This information is used to complete the purchase transaction.
Twisted VoIP may also collect anonymous demographic information, which is not unique to you, such as your:
Age
Gender
Geographic Location
Please keep in mind that if you directly disclose personally identifiable information or personally sensitive data through Twisted VoIP’s public message boards, this information may be collected and used by others.
We do not collect any personal information about you unless you voluntarily provide it to us. However, you may be required to provide certain personal information to us when you elect to use certain products or services. These may include: (a) registering for an account; (b) entering a sweepstake or contest sponsored by one of our partners or us; (c) signing up for special offers from selected third parties; (d) sending us an email message; (e) submitting your credit card or other payment information when ordering and purchasing products and services. To wit, we will use your information for, but not limited to, communicating with you about services and products you have requested from us. We also may gather additional personal or non-personal information in the future.
Use of your Personal Information
Twisted VoIP collects and uses your personal information to operate and deliver your requested services.
Twisted VoIP may also use your personally identifiable information to inform you of other products or services available from Twisted VoIP and its affiliates.
Sharing Information with Third Parties
Twisted VoIP does not sell, rent or lease its customer lists to third parties.
From time to time, Twisted VoIP may contact you on behalf of external business partners about a particular offering that may be of interest to you. In those cases, your unique, personally identifiable information (e-mail, name, address, telephone number) is not transferred to the third party. Twisted VoIP may share data with trusted partners to help perform statistical analysis, send you email or postal mail, provide customer support, or arrange deliveries. All such third parties are prohibited from using your personal information except to provide these services to Twisted VoIP, and they are required to maintain the confidentiality of your data.
Twisted VoIP may disclose your personal information without notice if required to do so by law or in the good faith belief that such action is necessary to: (a) conform to the edicts of the law or comply with the legal process served on Twisted VoIP or the site; (b) protect and defend the rights or property of Twisted VoIP; and (c) act under exigent circumstances to protect the personal safety of users of Twisted VoIP, or the public.
Tracking User Behavior
Twisted VoIP may keep track of the websites and pages our users visit within Twisted VoIP to determine what Twisted VoIP services are the most popular. This data is used to deliver customized content and advertising within Twisted VoIP to customers whose behavior indicates that they are interested in a particular subject area.
_________________
Automatically Collected Information
Twisted VoIP may automatically collect information about your computer hardware and software. This information can include your IP address, browser type, domain names, access times, and referring website addresses. This information is used for the operation of the service, to maintain the quality of the service, and to provide general statistics regarding the use of the Twisted VoIP website.
Use of Cookies
The Twisted VoIP website may use “cookies” to help you personalize your online experience. A cookie is a text file placed on your hard disk by a web page server. Cookies cannot run programs or deliver viruses to your computer. Cookies are uniquely assigned to you and can only be read by a web server in the domain that issued the cookie to you.
One of the primary purposes of cookies is to provide a convenience feature to save you time. The purpose of a cookie is to tell the Web server that you have returned to a specific page. For example, if you personalize Twisted VoIP pages or register with Twisted VoIP sites or services, a cookie helps Twisted VoIP recall your detailed information on subsequent visits. This simplifies the process of recording your personal information, such as billing addresses, shipping addresses, and so on. When you return to the same Twisted VoIP website, the previously provided information can be retrieved to use the Twisted VoIP features you customized efficiently.
You can accept or decline cookies. Most Web browsers automatically accept cookies, but you can usually modify your browser setting to decline cookies if you prefer. If you choose to decline cookies, you may not fully experience the interactive features of the Twisted VoIP services or websites you visit.
Links
This website contains links to other sites. Please be aware that we are not responsible for such other sites’ content or privacy practices. We encourage our users to be aware when they leave our site and to read the privacy statements of any other site that collects personally identifiable information.
Security of your Personal Information
Twisted VoIP secures your personal information from unauthorized access, use, or disclosure. Twisted VoIP uses the following methods for this purpose:
SSL Protocol
When personal information (such as a credit card number) is transmitted to other websites, it is protected through encryption, such as the Secure Sockets Layer (SSL) protocol.
We strive to take appropriate security measures to protect against unauthorized access to or alteration of your personal information. Unfortunately, no data transmission over the Internet or any wireless network can be 100% secure. As a result, while we strive to protect your personal information, you acknowledge that: (a) there are security and privacy limitations inherent to the Internet which are beyond our control; and (b) security, integrity, and privacy of any information and data exchanged between you and us through this Site cannot be guaranteed.
Right to Deletion
Subject to certain exceptions set out below, on receipt of a verifiable request from you, we will:
Delete your personal information from our records; and
Direct any service providers to delete your personal information from their records.
Please note that we may not be able to comply with requests to delete your personal information if it is necessary to:
Complete the transaction for which the personal data was collected, fulfill the terms of a written warranty or product recall conducted by federal law, provide a good or service requested by you or reasonably anticipated within the context of our ongoing business relationship with you, or otherwise perform a contract between you and us;
Detect security incidents, protect against malicious, deceptive, fraudulent, or illegal activity; or prosecute those responsible for that activity;
Debug to identify and repair errors that impair existing intended functionality;
Exercise free speech, ensure the right of another consumer to exercise their right of free speech, or exercise another right provided for by law;
Comply with the California Electronic Communications Privacy Act;
Engage in public or peer-reviewed scientific, historical, or statistical research in the public interest that adheres to all other applicable ethics and privacy laws when our deletion of the information is likely to render impossible or seriously impair the achievement of such research, provided we have obtained your informed consent;
Enable solely internal uses that are reasonably aligned with your expectations based on your relationship with us;
Comply with an existing legal obligation; or
Otherwise, use your personal information internally, in a lawful manner compatible with the context in which you provided the information.
Children Under Thirteen
Twisted VoIP does not knowingly collect personally identifiable information from children under thirteen. If you are under the age of thirteen, you must ask your parent or guardian for permission to use this website.
Disconnecting your Twisted VoIP Account from Third Party Websites
You will be able to connect your Twisted VoIP account to third-party accounts. BY CONNECTING YOUR TWISTED VOIP ACCOUNT TO YOUR THIRD-PARTY ACCOUNT, YOU ACKNOWLEDGE AND AGREE THAT YOU ARE CONSENTING TO THE CONTINUOUS RELEASE OF INFORMATION ABOUT YOU TO OTHERS (BY YOUR PRIVACY SETTINGS ON THOSE THIRD-PARTY SITES). IF YOU DO NOT WANT INFORMATION ABOUT YOURSELF, INCLUDING PERSONALLY IDENTIFYING INFORMATION, TO BE SHARED IN THIS MANNER, DO NOT USE THIS FEATURE. You may disconnect your account from a third-party account at any time. Users may learn to disconnect their accounts from third-party websites by visiting their “My Account” page. Users may also contact us via email or telephone.
E-mail Communications
From time to time, Twisted VoIP may contact you via email to provide announcements, promotional offers, alerts, confirmations, surveys, and other general communication. To improve our Services, we may receive a notification when you open an email from Twisted VoIP or click on a link therein.
Suppose you would like to stop receiving marketing or promotional communications via email from Twisted VoIP. In that case, you may opt-out of such communications by “replying STOP” or “clicking on the UNSUBSCRIBE button…
External Data Storage Sites
We may store your data on servers provided by third-party hosting vendors we have contracted.
Changes to this Statement
Twisted VoIP reserves the right to change this Privacy Policy from time to time. We will notify you about significant changes in how we treat personal information by sending a notice to the primary email address specified in your account, placing a prominent notice on our website, and updating any privacy information. Your continued use of the Website and Services available after such modifications will constitute your: (a) acknowledgment of the modified Privacy Policy; and (b) agreement to abide and be bound by that Policy.
Contact Information
Twisted VoIP welcomes your questions or comments regarding this Statement of Privacy. If you believe that Twisted VoIP has not adhered to this Statement, please get in touch with Twisted VoIP at:
Twisted VoIP LLC
2942 N 24th Street, Suite 114
Phoenix, Arizona 85016
At Twisted VoIP LLC, we strive to provide exceptional service and products to our customers. We understand that occasionally, circumstances may require a return or refund. Please read our return policy carefully to understand your options and responsibilities regarding returns and refunds.
1. Unused Balances:
Only unused balances are eligible for a refund.
Month-to-month accounts are not eligible for a refund; they will stop working at the end of the monthly term if you choose to cancel your account.
2. Hardware Returns:
Only unused hardware will be accepted with prior authorization (RMA – Return Merchandise Authorization).
An RMA is required before any refund will be processed on ANY hardware.
An RA (Return Authorization) is required for unused balance returns on telecommunication services.
The sale of open-box or refurbished hardware is rare, but these items are not eligible for return or refund under any circumstance other than in the event of an error by Twisted VoIP LLC.
3. Software Purchases:
Purchased software that is licensed directly to the end user is not eligible for a refund.
4. Call Quality Issues:
An RA will not be issued due to call quality issues determined by Twisted VoIP LLC to be on the end-user side. This includes improper network configurations or internet issues.
Refund Process:
Please get in touch with our
customer support
team to initiate a refund request for unused balances or authorized hardware returns.
Provide necessary details, including your account information, invoice number, and a brief reason for the refund request.
Once the return is approved, follow the instructions provided by our customer support team.
Refunds will be processed within 7-10 business days after the returned items are received and inspected.
Important Notes:
Refunds will be issued using the original payment method.
Twisted VoIP LLC reserves the right to deny any refund request that does not meet our return policy criteria.
Customers are responsible for all shipping costs associated with returning hardware items.
Twisted VoIP LLC is not responsible for items damaged or lost in transit during the return process.
We appreciate your understanding and cooperation. Please get in touch with our
customer support
team if you have any questions about our return policy.
Twisted VoIP LLC reserves the right to revise, amend, or modify this policy at any time without prior notice.
The HT801/HT802 analog telephone adaptors provide transparent connectivity for analog phones and faxes to the world of internet voice. Connecting to any analog phone, fax or PBX, the HT801/HT802 are an effective and flexible solution for accessing internet-based telephone services and corporate intranet systems across established LAN and internet connections. The Grandstream handy tones HT801/HT802 are new additions to the popular handy tone ATA product family. This manual will help you learn how to operate and manage your HT801/HT802 analog telephone adaptor and make the best use of its many upgraded features including simple and quick installation, 3-way conferencing, direct IP-IP Calling, and new provisioning support among other features. The HT801/HT802 are very easy to manage and configure and are specifically designed to be an easy-to-use and affordable VoIP solution for both the residential user and the teleworker.
The HT801 is a one-port analog telephone adapter (ATA) while the HT802 is a 2-port analog telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution for residential and office environments. Its ultra-compact size, voice quality, advanced VoIP functionality, security protection and auto provisioning options enable users to take advantage of VoIP on analog phones and enables service providers to offer high quality IP service. The HT801/HT802 are an ideal ATA for individual use and for large scale commercial IP voice deployments.
The following table contains major features of HT801 and HT802:
HT801
HT802
1 SIP profile through 1 FXS port on HT801, 2 SIP profiles through 2 FXS ports on HT802,
single 10/100Mbps port on both models.
Support 3-way voice conferencing.
Wide range of caller ID formats. Advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multi-language prompts, flexible dial plan, and more.
T.38 Fax for creating Fax-over-IP and GR-909 Line Testing Functionalities.
TLS and SRTP security encryption technology to protect calls and accounts.
Automated provisioning options include TR-069 and XML config files.
Failover SIP server automatically switches to secondary server if main server loses connection.
Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning.
The following table resume all the technical specifications including the protocols / standards supported, voice codecs, telephony features, languages and Upgrade/ Provisioning settings for the HT801/HT802.
HT801
HT802
Interfaces
Telephone Interfaces
One (1) RJ11 FXS port
Two (2) RJ11 FXS ports
Network Interface
One (1) 10/100Mbps auto-sensing Ethernet port (RJ45)
LED Indicators
POWER, INTERNET, PHONE
POWER, INTERNET, PHONE1,
PHONE2
Factory Reset Button
Yes
Voice, Fax, Modem
Telephony Features
Caller ID display or block, call waiting, flash, blind or attended transfer, forward, hold, do not disturb, 3-way conference.
Voice Codecs
G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.722, G.723.1, G.729A/B, G.726-32, iLBC, OPUS, dynamic jitter buffer, advanced line echo cancellation
Fax over IP
T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-through.
Short/Long Haul Ring Load
5 REN: Up to 1km on 24 AWG
2 REN: Up to 1km on 24 AWG
Caller ID
Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID.
This chapter provides basic installation instructions including the list of the packaging contents and information for obtaining the best performance with the HT801/HT802.
The HT801 and HT802 are designed for easy configuration and easy installation, to connect your HT801 or HT802, please follow the steps above:
Insert a standard RJ11 telephone cable into the phone port and connect the other end of the telephone cable to a standard touch-tone analog telephone.
Insert the Ethernet cable into the internet or LAN port of the HT801/ht802 and connect the other end of the Ethernet cable to an uplink port (a router or a modem, etc.)
Insert the power adapter into the HT801/HT802 and connect it to a wall outlet.
Power, Ethernet and Phone LEDs will be solidly lit when the HT801/HT802 is ready for use.
There are 3 LED buttons on HT801 and 4 LED buttons on HT802 that help you manage the status of your Handy Tone.
HT801/HT802 LEDs Pattern
LED Lights
Status
Power LED
The Power LED lights up when the HT801/HT802 is powered on, and it flashes when the HT802 is booting up
WAN LED
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.
Phone LED for HT801
Phone LED 1&2 for HT802
The phone LEDs
1 & 2 indicate the status of the respective FXS port-phone on the back panel
The HT801/HT802 can be configured via one of two ways:
The IVR voice prompt menu.
The Web GUI embedded on the HT801/HT802 using PC’s web browser.
Obtain HT80x IP Address via Connected Analog Phone #
HT801/HT802 is by default configured to obtain the IP address from DHCP server where the unit is located. To know which IP address is assigned to your HT801/HT802, you should access to the “Interactive Voice Response Menu” of your adapter via the connected phone and check its IP address mode.
Please refer to the steps below to access the interactive voice response menu:
Use a telephone connected to phone for the HT801 or phone 1 or phone 2 ports of your HT802.
Press *** (press the star key three times) to access the IVR menu and wait until you hear “Enter the menu option “.
Press 02 and the current IP address will be announced.
Understanding HT80x Interactive Voice Prompt Response Menu #
The HT801/HT802 has a built-in voice prompt menu for simple device configuration which lists actions, commands, menu choices, and descriptions. The IVR menu work with any phone connected to the HT801/HT802. Pick up the handset and dial “***” to use the IVR menu.
Menu
Voice Prompt
Options
Main Menu
“Enter a Menu Option”
Press “*” for the next menu option
Press “#” to return to the main menu
Enter 01-05, 07,10, 12-17,47 or 99 menu options
01
“DHCP Mode”,
“Static IP Mode”
“PPPoE Mode“
Press “9” to toggle the selection
If using “Static IP Mode”, configure the IP address information using menus 02 to 05.
If using “Dynamic IP Mode”, all IP address information comes from the DHCP server automatically after reboot.
If using “PPPoE Mode”, configure PPPoE Username and Password from web GUI to get IP from your ISP.
02
“IP Address “ + IP address
The current WAN IP address is announced If using “Static IP Mode”, enter 12-digit new IP address.
You need to reboot your HT812/HT814 for the new IP address to take Effect.
03
“Subnet “ + IP address
Same as menu 02
04
“Gateway “ + IP address
Same as menu 02
05
“DNS Server “ + IP address
Same as menu 02
07
Preferred Vocoder
Press “9” to move to the next selection in the list:
PCM U / PCM A
iLBC
G-726
G-723
G-729
OPUS
G722
10
“MAC Address”
Announces the MAC address of the unit.
Note: The device has two MAC addresses. One for the WAN port and one for the LAN port. The device MAC address announced is the address of LAN port.
12
WAN Port Web Access
Press “9” to toggle between enable / disable.
Default is disabled.
13
Firmware Server IP Address
Announces current Firmware Server IP address. Enter 12-digit new IP address.
14
Configuration Server IP Address
Announces current Config Server Path IP address. Enter 12-digit new IP address.
15
Upgrade Protocol
Upgrade protocol for firmware and configuration update. Press “9” to toggle between TFTP/HTTP/HTTP /FTP/FTPS
16
Firmware Version
Announces Firmware version information.
17
Firmware Upgrade
Firmware upgrade mode. Press “9” to toggle among the following three options:
Always check
Check when pre/suffix changes
Never upgrade
47
“Direct IP Calling”
Enter the target IP address to make a direct IP call, after dial tone.
86
Voice Mail
Access to your voice mails messages.
99
“RESET”
Press “9” to reboot the device
Enter MAC address to restore factory default setting
(See Restore Factory Default Setting section)
“Invalid Entry”
Automatically returns to main menu
“Device not registered”
This prompt will be played immediately after off hook If the device is not registered and the option “Outgoing Call without Registration” is in NO
Five success tips when using the voice prompt
“*” shifts down to the next menu option and “#” returns to the main menu.
“9” functions as the ENTER key in many cases to confirm or toggle an option.
All entered digit sequences have known lengths – 2 digits for menu option and 12 digits for IP address. For IP address, add 0 before the digits if the digits are less than 3 (i.e. – 192.168.0.26 should be key in like 192168000026. No decimal is needed).
Key entry cannot be deleted but the phone may prompt error once it is detected.
Dial *98 to announce the extension number of the port.
The HT801/HT802 embedded Web server responds to HTTP GET/POST requests. Embedded HTML pages allow a user to configure the HT801/HT802 through a web browser such as Google Chrome, Mozilla Firefox and Microsoft’s IE.
Enter the HT801/HT802’s IP address in the address bar of the browser.
Enter the administrator’s password to access the Web Configuration Menu.
Notes:
The computer must be connected to the same sub-network as the HT801/HT802. This can be easily done by connecting the computer to the same hub or switch as the HT801/HT802.
Recommended Web browsers:
Microsoft Internet Explorer: version 10 or higher.
There are two default passwords for the login page:
User Level
User
Password
Web Pages Allowed
End User Level
user
123
Only Status and Basic Settings
Administrator Level
admin
admin
Browse all pages
Viewer Level
viewer
viewer
View all pages. No changes allowed.
The password is case-sensitive and must contain 8-20 characters, at least one number, one uppercase, and one lowercase letter, When changing any settings, always submit them by pressing the Update or Apply button on the bottom of the page. After submitting the changes in all the Web GUI pages, reboot HT801/HT802 to have the changes take effect if necessary; most of the options under the AdvancedSettings and FXSPort(x) pages require a reboot.
After users make changes to the configuration, pressing the Update button will save but not apply the changes until the Apply button is clicked. Users can instead directly press the Apply button. We recommend rebooting or powering cycle the phone after applying all the changes.
Access your HT801/HT802 web UI by entering its IP address in your favorite browser.
Enter your admin password (default: admin).
Press Login to access your settings.
Go to Basic Settings 🡪 New End User Password and enter the new end-user password. (Must contain 8-20 characters, at least one number, one uppercase, and one lowercase letter.)
Confirm the new end-user password.
Press Apply at the bottom of the page to save your new settings.
Access your HT801/HT802 web UI by entering its IP address in your favorite browser.
Enter your admin password (default: admin).
Press Login to access your settings.
Go to Basic Settings 🡪 New Viewer Password and enter the new viewer password. (Must contain 8-20 characters, at least one number, one uppercase, and one lowercase letter.)
Confirm the new viewer password.
Press Apply at the bottom of the page to save your new settings.
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:
STUN Server (under advanced settings webpage) Enter a STUN server IP (or FQDN) that you may have or look up a free public STUN Server on the internet and enter it on this field. If using Public IP, keep this field blank.
Use Random SIP/RTP Ports (under advanced settings webpage) This setting depends on your network settings. Generally, if you have multiple IP devices under the same network, it should be set to Yes. If using a public IP address, set this parameter to No.
NAT traversal (under the FXS web page) Set this to Yes when gateway is behind firewall on a private network.
Select voice menu option 01 to allow the HT801/HT802 to use DHCP.
STATIC IP MODE
Select voice menu option 01 to allow the HT801/HT802 to enable the STATIC IP mode, then use option 02, 03, 04, 05 to set up IP address, Subnet Mask, Gateway and DNS server respectively.
FIRMWARE SERVER IP ADDRESS
Select voice menu option 13 to configure the IP address of the firmware server.
CONFIGURATION SERVER IP ADDRESS
Select voice menu option 14 to configure the IP address of the configuration server.
UPGRADE PROTOCOL
Select the menu option 15 to choose firmware and configuration upgrade protocol between TFTP, HTTP and HTTPS, FTP and FTPS. Default is HTTPS.
FIRMWARE UPGRADE MODE
Select voice menu option 17 to choose firmware upgrade mode among the following three options:
“Always check, check when pre/suffix changes, and never upgrade”.
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.
Access your HT801/HT802 web UI by entering its IP address in your favorite browser.
Enter your admin password (default: admin) and press Login to access your settings.
Go to FXS Port (1 or 2) pages.
In FXS Port tab, set the following:
Account Active to Yes.
Primary SIP Server field with your SIP server IP address or FQDN.
Failover SIP Server with your Failover SIP Server IP address or FQDN. Leave empty if not available.
Prefer Primary SIP Server to No or Yes depending on your configuration. Set to No if no Failover SIP Server is defined. If “Yes”, account will register to Primary SIP Server when failover registration expires.
Outbound Proxy: Set your Outbound Proxy IP Address or FQDN. Leave empty if not available.
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.
Authenticate ID: SIP service subscriber’s Authenticate ID used for authentication. Can be identical to or different from SIP User ID.
Authenticate Password: SIP service subscriber’s account password to register to SIP server of ITSP. For security reasons, the password will field will be shown as empty.
Name: Any name to identify this specific user.
Press Apply at the bottom of the page to save your configuration.
SIP Accounts Settings
After applying your configuration, your account will register to your SIP Server, you can verify if it has been correctly registered with your SIP server or from your HT801/HT802 web interface under Status > Port Status > Registration (If it displays Registered, it means that your account is fully registered, otherwise it will display NotRegistered so in this case you must double check the settings or contact your provider).
Account Status
When all the FXS ports are registered (for HT802), the simultaneous ring will have one second delay between each ring on each phone.
Press the “Reboot” button at the bottom of the configuration menu to reboot the ATA remotely. The web browser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log in again.
Direct IP Calling. Dial “*47” + “IP address”. No dial tone is played in the middle.
*50
Disable Call Waiting (for all subsequent calls)
*51
Enable Call Waiting (for all subsequent calls)
*67
Block Caller ID (per call). Dial “*67” +” number”. No dial tone is played in the middle.
*82
Send Caller ID (per call). Dial “*82” +” number”. No dial tone is played in the middle.
*69
Call Return Service: Dial *69 and the phone will dial the last incoming phone number received.
*70
Disable Call Waiting (per call). Dial “*70” +” number”. No dial tone is played in the middle.
*71
Enable Call Waiting (per call). Dial “*71” +” number”. No dial tone is played in the middle
*72
Unconditional Call Forward: Dial “*72” and then the forwarding number followed by “#”. Wait for dial tone and hang up. (dial tone indicates successful forward)
*73
Cancel Unconditional Call Forward. To cancel “Unconditional Call Forward”, dial “*73”, wait for dial tone, then hang up.
*74
Enable Paging Call: Dial “*74” and then the destination phone number you want to page.
*78
Enable Do Not Disturb (DND): When enabled all incoming calls are rejected.
*79
Disable Do Not Disturb (DND): When disabled, incoming calls are accepted.
*87
Blind Transfer.
*90
Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for dial tone then hang up.
*91
Cancel Busy Call Forward. To cancel “Busy Call Forward”, dial “*91”, wait for dial tone, then hang up.
*92
Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for dial tone then hang up.
*93
Cancel Delayed Call Forward. To cancel Delayed Call Forward, dial “*93”, wait for dial tone, then hang up
Flash/Hook
Toggles between active call and incoming call (call waiting tone). If not in conversation, flash/hook will switch to a new channel for a new call.
To make the outgoing calls using your HT801/HT802:
Pick up the handset of the connected phone;
Dial the number directly and wait for 4 seconds (Default “No Key Entry Timeout”); or
Dial the number directly and press # (Use # as dial key” must be configured in web configuration).
Examples:
Dial an extension directly on the same proxy, (e.g. 1008), and then press the # or wait for 4 seconds;
Dial an outside number (e.g. (626) 666-7890), first enter the prefix number (usually 1+ or international code) followed by the phone number. Press # or wait for 4 seconds. Check with your VoIP service provider for further details on prefix numbers.
Notes:
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:
• Set “Play Busy Tone When Account is unregistered” to YES under Advanced Settings.
• Set “Outgoing call without registration” to NO under FXS Port (1,2).
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.
Elements necessary to completing a Direct IP Call:
Both HT801/HT802 and other VoIP Device, have public IP addresses, or
Both HT801/HT802 and other VoIP Device are on the same LAN using private IP addresses, or
Both HT801/HT802 and other VoIP Device can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ).
The HT801/HT802 supports two ways to make Direct IP Calling:
Using IVR
Pick up the analog phone then access the voice menu prompt by dialing “***”;
Dial “47” to access the direct IP call menu;
Enter the IP address after the dial tone and voice prompt “Direct IP Calling”.
Using Star Code
Pick up the analog phone then dial “*47”;
Enter the target IP address.
No dial tone will be played between step 1 and 2 and destination ports can be specified using “*” (encoding for “:”) followed by the port number.
Examples of Direct IP Calls:
a) If the target IP address is 192.168.0.160, the dialing convention is *47 or Voice Prompt with option 47, then 192*168*0*160, followed by pressing the “#” key if it is configured as a send key or wait 4 seconds. In this case, the default destination port 5060 is used if no port is specified;
b) If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be: *47 or Voice Prompt with option 47, then 192*168*0*160*5062 followed by pressing the “#” key if it is configured as a send key or wait for 4 seconds.
When completing direct IP call, the “Use Random SIP/RTP Port” should be set to “NO”.
You can place a call on hold by pressing the “flash” button on the analog phone (if the phone has that button).
Press the “flash” button again to release the previously held Caller and resume conversation. If no “flash” button is available, use “hook flash” (toggle on-off hook quickly). You may drop a call using hook flash.
Assume that the call is established between phone A and B are in conversation. The phone A wants to attend transfer phone B to phone C:
On the phone A presses FLASH to hear the dial tone.
Phone A dials the phone C’s number followed by # (or wait for 4 seconds).
If phone C answers the call, phones A and C are in conversation. Then A can hang up to complete transfer.
If phone C does not answer the call, phone A can press “flash” to resume call with phone B.
When attended transfer fails and A hangs up, the HT801/HT802 will ring back user A to remind A that B is still on the call. A can pick up the phone to resume conversation with B.
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:
Phone A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone.
Phone A dials C’s number then # (or wait for 4 seconds).
If phone C answers the call, then A presses FLASH to bring B, C in the conference.
If phone C does not answer the call, phone A can press FLASH back to talk to phone B.
If phone A presses FLASH during conference, the phone C will be dropped out.
If phone A hangs up, the conference will be terminated for all three parties when configuration “Transfer on Conference Hang up” is set to “No”. If the configuration is set to “Yes”, A will transfer B to C so that B and C can continue the conversation.
Pick up the handset of the connected phone (Off-hook).
After hearing the dial tone, input “*69”.
Your phone will automatically call back to the latest incoming number.
All star code (*XX) related features mentioned above are supported by ATA default settings. If your service provider provides different feature codes, please contact them for instructions.
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.
On the HT802 inter-port calling is achieved by dialing ***7X (X is the port number). For example, the user connected to port 1 can be reached by dialing *** and 71.
Warning: Restoring the Factory Default Settings will delete all configuration information on the phone. Please backup or print all the settings before you restore to the factory default settings. Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
There are three (3) methods for resetting your unit:
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.
Firmware Version 1.0.53.2
No Major Changes.
Firmware Version 1.0.51.1
No Major Changes.
Firmware Version 1.0.49.2
No Major Changes.
Firmware Version 1.0.47.4
Added IVR support to check Device Individual Certificate Information. [Interactive Voice Prompt Response Menu]
Firmware Version 1.0.45.2
No Major Changes.
Firmware Version 1.0.43.11
Added Charter CA to the approved certificate list.
