HT70X USER MANUAL INDEX GNU GPL INFORMATION ................. 5 CHANGE LOG ................... 6 CHANGES FROM 18.104.22.168 USER MANUAL ..................6 CHANGES FROM 22.214.171.124 USER MANUAL ..................6 CHANGES FROM 126.96.36.199 USER MANUAL ..................6 WELCOME ....................7 SAFETY COMPLIANCES ........................7 WARRANTY ............................
CONFIGURING THE HT70X VIA WEB BROWSER ................23 Access the Web Configuration Menu ................... 24 IMPORTANT SETTINGS ........................24 NAT Settings ..........................24 DTMF Methods ..........................25 Preferred VOCODER (Codec) ...................... 25 SAVING THE CONFIGURATION CHANGES ..................44 REBOOTING THE HT70X FROM REMOTE ..................44 CONFIGURATION THROUGH A CENTRAL SERVER ..............
ABLE CCOUNT ETTINGS 12: HT704 FXS P ...................... 43 ABLE ORTS ETTINGS GUI I ONFIGURATION NTERFACE XAMPLES HT70X USER MANUAL (http://www.grandstream.com/products/ht_series/ht701/documents/ht70x_gui.zip) 1. S CREENSHOT OF DVANCED ONFIGURATION 2. S CREENSHOT OF ASIC ETTINGS ONFIGURATION 3. S FXS P CREENSHOT OF ONFIGURATION 4.
HT70X firmware contains third-party software licensed under the GNU General Public License (GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public License (GPL) for the exact terms and conditions of the license.
CHANGE LOG This section documents significant changes from previous versions of HT70X user manuals. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here. CHANGES FROM 188.8.131.52 USER MANUAL ...
Thank you for purchasing Grandstream’s HT70X, the affordable, feature rich Analog Telephone Adaptor. Grandstream HandyTone70X is a new addition to the popular HandyTone ATA product family. It features the rich audio quality, a broad range of voice codecs, and functionality including one independent SIP account per FXS port.
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose is not permitted without the express written permission of Grandstream Networks, Inc. FIRMWARE VERSION 184.108.40.206 HT70X USER MANUAL...
CONNECT YOUR HT70X Connecting the HT70X is easy. Before you begin, please verify the contents of the HT70X package. EQUIPMENT PACKAGING The HT70X ATA package contains: One HT70X Main Case One Universal Power Adaptor One Ethernet Cable ...
HT-701 Phone (RJ-11 FXS Ports) Internet Port Power (RJ-45 connector Display LEDs Supply 10/100 Mbps) Reset (green) (12V;0.5A) HT702 HT704 Display LED’s (green) Display LED’s (green) LAN Port Power LAN Port Phone (RJ-11 Power Phone (RJ-11 Reset Reset (RJ-45 connector Supply (RJ-45 connector Supply...
PHONE LED Indicate status of the respective FXS Ports-PHONE on the back panel Unregistered – OFF Registered and Available – ON (Solid Green) Off-Hook / Busy – Blinking every second Slow blinking FXS LEDs indicates voicemail NOTE: All LEDs display green when ON TABLE 3: ADVANCED DEFINITIONS OF THE HT70X LEDS PATTERN LED-01 Device has normal power...
3sec OFF LED-13 Receiver off hook test fail. One or more phones are off 4x250ms ON/OFF hook on phone line during test. Phone 3sec OFF REN test failed – high REN detected. Too many parallel LED-14 5x250ms ON/OFF phones connected to phone line X Phone 3sec OFF LED-15...
HT70X FEATRUES The HT70X is a full feature voice and fax-over IP device that offers a high-level of integration including a 10M/100Mbps network port and one FXS telephone port, market-leading sound quality, rich functionalities, and a compact and lightweight design. The VoIP network signaling protocol supported is SIP.
Device Management Web interface or via secure encrypted AES or non-encrypted central configuration file for mass deployment using Grandstream binary file or xml format. Auto/manual provisioning system or via built-in IVR. NAT-friendly remote software upgrade (via TFTP/HTTP/HTTPS) for deployed devices including behind firewall/NAT.
BASIC OPERATIONS UNDERSTANDING HT70X VOICE PROMPT HT70X has a built-in voice prompt menu for simple device configuration. The IVR menu and the LED button work with any of the FXS port. Pick up the handset and dial “***” to use the IVR menu. TABLE 6: HT70X IVR MENU DEFINITIONS MENU VOICE PROMPT...
toggle between TFTP / HTTP / HTTPS Firmware Version Firmware version information. Firmware upgrade mode. Press “9” to toggle among the following Firmware Upgrade three options: - always check - check when pre/suffix changes - never upgrade “Direct IP Calling” Enter the target IP address to make a direct IP call, after dial tone.
PLACING A PHONE CALL PHONE OR EXTENSION NUMBERS 1. Dial the number directly and wait for 4 seconds (Default “No Key Entry Timeout”); or 2. Dial the number directly and press # (Use # as dial key” must be configured in web configuration). Examples: 1.
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 “#”...
ATTENDED TRANSFER Assume that Caller A and B are in conversation. Caller A wants to Attend Transfer B to C: 1. Caller A presses FLASH on the analog phone for dial tone. 2. Caller A then dials Caller C’s number followed by # (or wait for 4 seconds). 3.
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. Pressing pound sign will serve as Re-Dial key.
CONFIGURATION GUIDE CONFIGURING THE HT70X THROUGH VOICE PROMPTS DHCP M Select voice menu option 01 to enable HT70X to use DHCP. STATIC IP M Select voice menu option 01 to enable HT70X to use STATIC IP mode, then use option 02, 03, 04, 05 to set up IP address, Subnet Mask, Gateway and DNS server respectively.
“ADVANCED SETTING”, “FXS PORTs” configuration pages. Please reference the GUI pages using the following link: http://www.grandstream.com/products/ht_series/ht701/documents/ht70x_gui.zip NOTE: If you cannot log into the configuration page by using the default password, please check with the VoIP service provider. It is most likely the VoIP service provider has provisioned the device and configured for you therefore the password has already been changed.
1. 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. 2.
Telnet Server Default is set to Yes. HTTP Access Default is set to Yes. If set to No, http access will be denied. IP Address There are two modes to operate the HT70X: DHCP mode: all the field values for the Static IP mode are not used (even though they are still saved in the Flash memory.) The HT701 acquires its IP address from the first DHCP server it discovers from the LAN it is connected.
Self-Defined Time Zone The syntax is std offset dst [offset],start[/time],end[/time] Default is set to : MTZ+6MDT+5,M3.2.0,M11.1.0 MTZ+6MDT+5, Time zone with 6 hours offset with 1 hour ahead which is the US central time. It is positive (+) if the local time zone is west of the Prime Meridian and negative (-) if it is east. Prime Meridian (a.k.a: International or Greenwich Meridian) M3.2.0,M11.1.0 The 1...
Product Model This field contains the product model info. Hardware Version This field shows the hardware revision of the unit and the part number. Software Version Program: This is the main software release. This number is always used for firmware upgrade.
TABLE 10: ADVANCED SETTINGS Admin Password This contains the password to access the Advanced Web Configuration page. This field is case sensitive. Only the administrator can configure the “Advanced Settings” page. Password field is purposely left blank for security reasons after clicking update and saved. The maximum password length is 25 characters.
The URL for the HTTP/HTTPS server used for firmware upgrade and configuration via Server HTTP. For example, http://provisioning.mycompany.com:6688/Grandstream/220.127.116.11“:6688” is the specific TCP port where the HTTP or HTTPS server is listening; it can be omitted if using default port 80.
SIP TLS Certificate The user specify SSL certificate used for SIP over TLS in X.509 format. SIP TLS Private Key The user specify SSL private key used for SIP over TLS in X.509 format. SIP TLS Private Key User specify password to protect the private key above. Password User specify the Auto Configuration Server’s URL (TR-069 protocol) ACS URL...
Default is No. If set to “Yes”, the configuration update via keypad is disabled. Lock Keypad Update Disable Voice Prompt Default is No. Disables the voice prompt configuration. Disable Direct IP Call Default is No. Disables the Direct IP Call function. Failover to FXO Default is Disable.
Syslog Level Select the HT701 to report the log level. Default is NONE. The level is one of EXTRA DEBUG, DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the following events: 1. product model/version on boot up (INFO level) 2.
Default is 1813. Specifies the port to be used for the Primary RADIUS Account. HT704 Primari RADIUS Acct Only Port Specifies the secret string to be used to authenticate the RADIUS connection to the Primary RADIUS Primary Server. It should match RADIUS configuration. HT704 Only Server Secret Set the IP or FQDN of the Secondary RADIUS Server.
environments. If symmetric NAT is detected, STUN will not work and ONLY outbound proxy can correct the problem. SIP transport User can select UDP or TCP or TLS. Default is UDP. NAT Traversal (STUN) This parameter defines whether or not the HT70X NAT traversal mechanism is activated.
calls. Register Expiration This parameter allows the user to specify the time frequency (in minutes) the HT70X refreshes its registration with the specified registrar. The default interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 days). Reregister before Default is 0 (function disabled).
Validate incoming SIP Default is No. If set to yes all incoming SIP messages will be strictly validated message according to RFC rules. If message will not pass validation process, call will be rejected. Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the Check SIP User ID for incoming INVITE call will be rejected.
the FXS port or Profile. Distinctive ring tones can be configured not only for matching a whole number, but also for matching prefixes. In this case symbol * (star) will be used. For example: if configured as *617, Ring Tone 1 will be used in case of call arrived from the area code 617.
No Key Entry Timeout Default is 4 seconds. Dialing process is completed and outgoing call is initiated if no key entry occurs during this preconfigured interval. Early Dial Default is No. Use only if proxy supports 484 response. This parameter controls whether the phone will send an early INVITE each time a key is pressed when a user dials a number.
Softswitch vendors. Session Expiration Grandstream implemented SIP Session Timer. The session timer extension enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or re-INVITE message, the session will be terminated.
session timer only when the remote party support this feature. To turn off Session Timer, select “No” for Caller Request Timer, Callee Request Timer, and Force Timer. UAC Specify Refresher Default is Omit. As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher.
Jitter Buffer Length Select Low, Medium or High based on network conditions. Default is Medium. High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the high requirement Medium (initial 100ms, min 20ms, max 200ms) ...
The configuration, completed in Distinctive Ring Tones block in the same page, applies to ring tones cadences configured here. TABLE 12: HT704 FXS PORTS SETTINGS SIP Use ID User account information, provided by VoIP service provider (ITSP). Usually in the form of digit similar to phone number or actually a phone number.
Map to FXP Port # This is used only when peering with a Grandstream GXW410x. Default is 1, Supported values are 1-8, meaning line 1 to line 8 of the GXW410x device where the port will be mapped to. Map to FXO Gateway IP This is used when peering with an FXO gateway of any brand.
TFTP or HTTP (thus no NAT issues) and other communication protocols to communicate with each individual Grandstream device for firmware upgrade, remote reboot, etc. Grandstream provides GAPS service to VoIP service providers. Use GAPS for either simple redirection or with certain special provisioning settings.
CONFIGURATION GUIDE section for instructions. If the server is in FQDN format, it must be set via the web configuration interface. Grandstream recommends end-user use the Grandstream HTTP server. Its address can be found at http://www.grandstream.com/support/firmware . Currently the HTTP firmware server IP address is firmware.grandstream.com.
For users who do not have a local firmware upgrade server, Grandstream provides a NAT-friendly TFTP server on the public Internet for firmware upgrade. Grandstream’s latest firmware is available at http://www.grandstream.com/support/firmware Oversea users are strongly recommended to download the binary files and upgrade firmware locally in a controlled LAN environment.
When a Grandstream device boots up or reboots, it will issue a request for a configuration file “cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the MAC address of the device, i.e., “cfg000b820102ab”. In addition, device will also requests a XML configuration file “cfgxxxxxxxxxxxx.xml”.
periodically check if there are any new changes need to be taken on a scheduled time. By defining different intervals in P193 for different devices, Server Provider can spread the Firmware or Configuration File download in minutes to reduce the Firmware or Provisioning Server load at any given time. Automatic Upgrade: Yes, every minutes(60-5256000).
Restoring the Factory Default Setting will DELETE all configuration information of the phone. Please BACKUP or PRINT out all the settings before you approach to following steps. Grandstream will not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP service provider.
2222 33 (press the “3” key twice, “D” will show on the LCD) 3333 For example: if the MAC address is 000b8200e395, it should be keyed in as “0002228200333395”. RESET FROM WEB INTERFACE (RESET TYPE) 1. From the Advanced Settings Page user can select three types: ...