Grandstream Networks, Inc.
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HT70X USER MANUAL INDEX GNU GPL INFORMATION ......................................................................... 4 CHANGE LOG ........................................................................................... 5 CHANGES FROM 1.0.7.3 USER MANUAL .......................................................................................... 5 CHANGES FROM 1.0.6.1 USER MANUAL .......................................................................................... 5 CHANGES FROM 1.0.5.2 USER MANUAL ..
CALL FEATURES .................................................................................... 25 CONFIGURATION GUIDE ....................................................................... 27 CONFIGURING THE HT70X THROUGH VOICE PROMPTS ............................................................ 27 CONFIGURING THE HT70X VIA WEB BROWSER ........................................................................... 27 Access the Web Configuration Menu ........................................................
TABLE OF FIGURES HT70X USER MANUAL FIGURE 1: CONNECTING THE HT70X .................................................................................................... 12 FIGURE 2: HT70X CONNECTION DIAGRAM .......................................................................................... 14 TABLE OF TABLES HT70X USER MANUAL TABLE 1: DEFINISIONS OF THE HT70X CONNECTORS ....................................................................................
GNU GPL INFORMATION 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. Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: http://www.grandstream.com/support/faq/gnu_gpl . FIRMWARE VERSION 1.0.8.
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 1.0.7.3 USER MANUAL Upgrade the option Allow DHCP Option 66 to override server to [Allow DHCP Option 66 or 160 Added the options to enable/disable [3CX Auto ProvisionDo Not Escape '#' as %23 in SIP URI].
Added the options to enable/disable [Generate Continuous RFC2833 Events]. Added the options to configure [SIP REGISTER Contact Header UsesDo Not Escape '#' as %23 in SIP URI] it’s WAN or LAN address. Added the options to [Disable Multiple m line in SDP]. Added the options to configure [SIP Timer D]. Added the options to configure [Country Specific Deployment]. CHANGES FROM 1.0.5.2 USER MANUAL Added the options to enable/disable [Always send HTTP Basic Authentication Information].
CHANGES FROM 1.0.0.
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WELCOME 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.
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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 One Vertical Stand (Only on HT702 and HT704 Packages) CONNECTING THE HT70X The HT70X is designed for easy configuration and easy installation. Configure the HT70X following the directions in the Configuration section of this manual. 1.
HT-701 Internet Port (RJ-45 connector 10/100 Mbps) Display LEDs (green) HT702 LAN Port (RJ-45 connector 10/100 Mbps with LEDS) Reset Reset Power Supply (12V;0.5A) HT704 Display LED’s (green) Phone (RJ-11 FXS Ports) Phone (RJ-11 FXS Ports) Power Supply (12V;1A) Display LED’s (green) Phone (RJ-11 FXS Ports) FIGURE 1: DIAGRAM OF HT70X TABLE 1: DEFINISIONS OF THE HT70X CONNECTORS DC 12V Power adapter connection Internet Port (RJ-45) Connect to the internal LAN network.
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 LED-02 Power Error: Power is removed from the device or power supply with improper voltage is plugged in LED-03 Line X is registered normally to the sip providers network and is ready to make a call LED-04 Voice mail waiting for Lin
LED-13 Receiver off hook test fail. One or more phones are off hook on phone line during test. 4x250ms ON/OFF Phone + 3sec OFF LED-14 REN test failed – high REN detected.
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. The HT70X is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market.
synchronization), AGC (automatic gain control), Status output, Decoder controlling via voice packet header DHCP Server/Client Yes, DHCP Client only Telnet Server Yes Fax over Ip T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Passthrough, Fax Datapump V.17, V.19, V.27ter, V.29 for T.38 fax relay QoS Diffserve, TOS, 802.
LED Indicators POWER, INTERNET, POWER, INTERNET, LINK/ACTIVITY, LINK/ACTIVITY, PHONE Factory Reset Button Universal Switching Power Adaptor Environmental PHONE1, PHONE2 POWER, PHONE1, PHONE2, PHONE3, PHONE4 Yes Input: 100–240 VAC/50-60 Hz 0.18A Max Output: 12VDC, 0.5A, UL certified Input: 100–240 VAC/50-60 Hz 0.
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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.
16 Firmware Version 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. (See “Make a Direct IP Call”.
Examples: 1. Dial an extension directly on the same proxy, (e.g. 1008), and then press the # or wait for 4 seconds. 2. 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.
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. NOTE: When completing direct IP call, the “Use Random SIP/RTP Port” should set to “NO”. You cannot make direct IP calls between FXS1 to FXS2 since they are using same IP.
NOTE: When Attended Transfer fails and A hangs up, the HT70X 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. 3-WAY CONFERENCING The HT701/702/704 supports Bellcore style 3-way Conference. Instructions for 3-way conference: Assume that call party A and B are in conversation. Caller A(HT70X) wants to bring third Caller C into conference: 1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone. 2.
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CALL FEATURES The HT70X supports all the traditional and advanced telephony features. TABLE 7: HT70X CALL FEATURES KEY *02 CALL FEATURES Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729), *0272616 (G726-r16), *0272624 (G724-r24), *0272632 (G726-r32), *0272640 (G726-r40), *027201 (iLBC) *03 Disable LEC (per call) Dial “*03” + ” number ”. No dial tone is played in the middle.
*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. # Pressing pound sign will serve as Re-Dial key. FIRMWARE VERSION 1.0.8.
CONFIGURATION GUIDE CONFIGURING THE HT70X THROUGH VOICE PROMPTS DHCP MODE Select voice menu option 01 to enable HT70X to use DHCP. STATIC IP MODE 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. FIRMWARE SERVER IP ADDRESS Select voice menu option 13 to configure the IP address of the firmware server.
Note: • The IVR announces 12 digits IP address, you need to strip out the leading “0” in the IP address. For ex. IP address: 192.168.001.014, you need to type in http://192.168.1.14 in the web browser. Once the HTTP request is entered and sent from a web browser, the user will see a log-in screen.
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. 3. NAT Traversal (under the FXS/Profile web page) Set this to Yes when gateway is behind firewall on a private network. DTMF METHODS DTMF Settings are in FXS portX/ProfileX page. DTMF in-audio DTMF via RTP (RFC2833) DTMF via SIP INFO Set priority of DTMF methods according to your preference.
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. Using the PPPoE feature: set the PPPoE account settings. The HT70x will establish a PPPoE session if any of the PPPoE fields is set.
Allow DHCP server to Default Yes. Let the DHCP server handle the Time Zone set Time Zone Language Languages supported with voice prompt and web interface, except Spanish that it is only in IVR. Reset Type Gives the user the option to set to default all VoIP related configuration (mainly everything located on FXS/Profile page), all ISP (Internet Service Provider) configuration which may affect the IP address, or both at the same time.
Port Status Displays relevant information regarding the FXS port. Port FXS Hook On Hook Registration Registered DND Forward Yes Busy Delayed Forward Forward 613 • FXS port is registered with SIP Server. • FXS Port user has set Do Not Disturb. • FXS Port user has set his calls to be unconditionally forwarded to ext 613. • FXS Port user has not set Busy or Delay call Forward. Log in to the advanced user configuration page the same way as for the basic configuration page.
Use STUN to detect Use STUN keep-alive to detect WAN side network problems. If keep-alive request does not network connectivity yield any response for configured number of times, the device will restart the TCP/IP stack. If the STUN server does not respond when the device boots up, the feature is disabled. Default setting is No. Use DNS to detect Use DNS to detect WAN side network problems. Default setting is Yes.
Always send HTTP Default is Send HTTP Basic Authentication Information only when challenged. If set to Basic Authentication Always, device will send configured user name and password within HTTP request without Information server sending authentication challenge. Firmware File Prefix Default is blank. If configured, HT701 will request firmware file with the prefix. This setting is useful for ITSPs. End user should keep it blank. Firmware File Postfix Default is blank. End user should keep it blank.
Periodic Inform Frequency that the inform packets will be sent out to the ACS Interval Connection Request Set a user name for the ACS to connect to this device Username Connection Request Set a password for the ACS to connect to this device Password CPE SSL Certificate The Cert File for the phone to connect to the ACS via SSL. CPE SSL Private Key The Cert Key for the phone to connect to the ACS via SSL. Country Specific Default is None.
FXO Gateway NTP server IP or Domain Name of the FXO Gateway that will be used as Failover. HT704 Only URI or IP address of the NTP (Network Time Protocol) server. This parameter synchronizes the date and time. Allow DHCP option 42 Default Yes. Enables the DHCP server to handle the NTP Server via Option 42 to override NTP serve Syslog Server The IP address or URL of System log server.
Set the IP or FQDN of the Primary RADIUS Server. HT704 Only Remote Authentication Dial In User Service (RADIUS). HT704 supports RADIUS for authentication, authorization and billing purposes. Primary and secondary RADIUS server Primary RADIUS Server configurations are available to provide redundancy to this feature. In case Primary Radius server becomes unusable, RADIUS requests will be automatically sent to the secondary server.
TABLE 11: ACCOUNT SETTINGS Profile/Account Active When set to Yes the FXS port or Profile (for HT704) is activated. Primary SIP Server SIP Server’s IP address or Domain name provided by VoIP service provider. Failover SIP Server Failover SIP Server’s IP address or Domain name in case primary server does not respond. Prefer Primary SIP Default is No.
Tel URI The default setting is “Disabled”. If the phone has an assigned PSTN Number, this field should be set to “User=Phone” then a “User=Phone” parameter will be attached to the “From header” in the SIP request to indicate the E.164 number. If server supports TEL URI format, then this option needs to be selected. SIP Registration Controls whether the HT701 needs to send REGISTER messages to the proxy server. The default setting is Yes. Unregister on Reboot Default is No.
The default value for FXS port is 5004. Use Random SIP Port Default is No. This parameter forces the random generation of The local SIP ports when set to Yes. This is usually necessary when multiple HT70X are behind the same NAT. Use Random RTP Port Default is No. This parameter forces the random generation of the local RTP ports when set to Yes. This is usually necessary when multiple HT70X are behind the same NAT. Hold Target Before Default is Yes. Allows user to hold the phone call before refer it.
Use P-PreferredIdentity Header SIP REGISTER Contact Header Uses SIP T1 Timeout If set to Default, it will add P-Preferred-Identity header unless special feature is Telkom SA or CBCOM. Default is LAN Address. If set to WAN Address, device will detect its WAN address and use it in SIP REGISTER Contact Header. T1 is an estimate of the round-trip time between the client and server transactions. If the network latency is high, select larger value for more reliable usage. Default is 0.5 Sec.
the Caller ID. System Ring Tone is used for all other calls. When selected but no Caller ID is configured, the selected ring tone will be used for all incoming calls using 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.
Delayed Call Forward Default value is 20 seconds. In case this feature activated using * codes (*92 code), the Wait Time call will be forwarded after this preconfigured amount of time. 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.
numbers with a minimum of 2 digits beginning with 2, with the leading digit replaced with 011. 3. Default: Outgoing - {x+} Example of a simple dial plan used in a Home/Office in the US: { ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 } Explanation of example rule (reading from left to right): • ^1900x.
Force Timer Default is No. If selecting “Yes” the phone will use session timer even if the remote party does not support this feature. Selecting “No” will allow the phone to enable 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.
Fax Mode T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA) Fax Tone Detection Default is Caller. This option is used to configure the mode how the device detect the Mode fax tone, it can be the mode of caller, callee and caller or callee. Re-Invite after Fax Tone Default is Enabled. It makes the unit send out the re-INVITE for T.38 or Fax Pass Detection Mode Through if a fax tone is detected.
User can adjust volume of call on either end using the Rx Gain Level parameter and the Tx Gain Level parameter located on the FXS Port Configuration page. If call volume is too low when using the FXS port (ie. the ATA is at user site), adjust volume using the Rx Gain Level parameter under the FXS Port Configuration page. If voice volume is too low at the other end, user may increase the far end volume using the Tx Gain Level parameter under the FXS Port Configuration page. Disable Line Echo Default is No.
For example: Port 1, 2, and 3 are members of the same Hunting Group. Port 1 is registered with a SIP account. Ports 2, and 3 are not registered. Ports 2 and 3 will be able to place outbound calls using the SIP account of port 1. Select appropriate value for Hunting Group feature. The original SIP account should be set to Active while the group members should be set to the port number of the Active Port.
When HT70X boots up, it will send TFTP or HTTP/HTTPS requests to download configuration files, “cfg000b82xxxxxx” and “cfg00082xxxxxx.xml”, where “000b82xxxxxx” is the LAN MAC address of the HT70X. If the download of “cfgxxxxxxxxxxxx.xml” is not successful, the provision program will issue request a generic configuration file “cfg.xml”. Configuration file name should be in lower case letters. The configuration data can be downloaded via TFTP or HTTP/HTTPS from the central server.
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SOFTWARE UPGRADE Software upgrade can be done via either TFTP or HTTP. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page. FIRMWARE UPGRADE THROUGH TFTP/HTTP/HTTPS To upgrade via TFTP or HTTP/HTTPS, the “Firmware Upgrade and Provisioning upgrade via” field needs to be set to TFTP HTTP or HTTPS, respectively. “Firmware Server Path” needs to be set to a valid URL of a TFTP or HTTP server, server name can be in either FQDN or IP address format.
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. Note: Alternatively, the user can upload the firmware single file from his/her computer using a local directory. For more instructions see below. Also the user can download a free TFTP or HTTP server and conduct local firmware upgrade.
It needs to be set to a valid URL, either in FQDN or IP address format. The “Config Server Path” can be same or different from the “Firmware Server Path”. A configuration parameter is associated with each particular field in the web configuration page. A parameter consists of a Capital letter P and 1 to 3 (Could be extended to 4 in the future) digit numeric numbers. i.e., P2 is associated with “Admin Password” in the ADVANCED SETTINGS page.
Automatic Upgrade: No Yes, every Yes, daily at hour FIRMWARE VERSION 1.0.8.2 minutes(60-5256000). (0-23). Yes, weekly on day HT70X USER MANUAL (0-6).
RESTORE FACTORY DEFAULT SETTING WARNING! 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. FACTORY RESET There are three (3) methods for resetting your unit: RESET BUTTON Reset default factory settings following these four (4) steps: 1.
D: 33 (press the “3” key twice, “D” will show on the LCD) E: 333 F: 3333 For example: if the MAC address is 000b8200e395, it should be keyed in as “0002228200333395”. RESET FROM WEB INTERFACE (RESET TYPE) 1.
