Thank you for purchasing Grandstream’s HT841/HT881. The HT841/HT881 Analog FXO Gateway series is an easy-to-set-up IP communications solution that is suitable for small enterprises, as well as those with virtual and branch locations, who wish to benefit from their broadband network or integrate new IP technology with their existing phone system.
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Operational: 32° – 104°F or 0º – 40ºC Environmental Storage: 14° – 140°F or -10º – 60ºC Humidity: 10 – 90% Non-condensing ● HT841/HT881 Dimensions and Weight Dimensions: 190mm x 100mm x 28mm (L x W x D) Weight: 0.46KG...
HT841/HT881 Ports Description Port Description DC 12V Power socket. Used to power HT841/HT881 (12V - 0.5A) NET 1/NET 2 Network NET 1 / NET 2 port. ● Used to power and connect your HT841/HT881 to local network when using as router.
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2. Connect the NET 1/ NET 2 port of the HT841/HT881 to a router, switch, or modem using an Ethernet cable. 3. Insert the power adapter into the HT841/HT881 and connect it to a wall outlet and make sure to respect the technical specifications of the power adapter used.
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HT841/HT881 LED Patterns LED Lights Status The Power LED lights up when the HT841/HT881 is powered on and it ashes when the HT841/HT88 is booting up. Power LED Note: When powering up the unit with the PSU, the LED light will glow green, whereas it will be orange when using PoE.
Server. Under the Channels webpage, you will need to fill in information like SIP User ID, Password, etc. Now, when you try to make calls from IP, the call will be routed to the SIP Server which will forward it to one of the SIP accounts on the HT841/HT881, which will then forward it to the PSTN line.
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FXS Gateway with HT841/HT881 [No SIP Server required] Alternatively, the HT841/HT881 can be used without a SIP Server. You can use it in conjunction with an FXS Gateway (Ex. GXW42xx) and still be able to originate and terminate calls from IP to PSTN and vice versa. All you need to make sure is that the 2 gateways are able to locate each other (they should be on the same LAN or on Public IP addresses).
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FXS port. FXO/FXS Lifeline Support *Refer to the following lifeline port info table: Model Lifeline Port Mapping HT841 FXO4 HT881 FXO8 The three modes of operation are as follows: Auto Mode (Default): In Auto mode, the Life Line feature will automatically activate when there is a power loss or when the device loses its registration with the SIP server (which means the VoIP service is unavailable).
Both HT841/HT881 and other VoIP Devices are on the same LAN using private IP addresses, or Both HT841/HT881 and other VoIP Devices can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ).
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The HT841/HT881 supports all the traditional and advanced telephony features, the following table summarizes each key with its corresponding Call Feature. Call Features Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729), *027201 (iLBC). Disable LEC (per call) Dial “*03” +” number”. No dial tone is played in the middle.
The embedded Web GUI on the HT841/HT881 using PC’s web browser. Obtain HT841/HT881 IP Address HT841/HT881 is by default configured to obtain the IP address from the DHCP server where the unit is located. To know which IP address is assigned to your HT841/HT881, you should access the “Interactive Voice Response...
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This prompt will be played immediately after off hook If the device is not registered and the option “Device not registered” “Outgoing Call without Registration” is in NO HT841/HT881 IVR Menu Four success tips when using the voice prompt “*” shifts down to the next menu option and “#” returns to the main menu “9”...
4. Enter the administrator’s password to access the Web Configuration Menu. The computer must be connected to the same sub-network as the HT841/HT881. This can be easily done by connecting the computer to the same hub or switch as the HT841/HT881.
Displays the memory usage System Up Time Displays the duration of the system up time CPU Load Displays the processor load Provision Displays the last status of provisioning System Info Downloads the system info Network Status MAC Address Displays the device's MAC address IPv4 Address Displays the assigned IPv4 address IPv6 Address...
Forward Indicates the forwarding number Busy Forward Indicates the busy forwarding number Delayed Forward Indicates the delayed forwarding number Indicates the Caller ID Call Waiting Indicates if Call waiting is enabled SRTP Indicates if Secured RTP is enabled System Settings Page Definitions Basic Settings The voice prompt is disabled if set to Yes.
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NTP Server This parameter sets the IP address of the NTP server. The device will obtain the date and time from the server Allow DHCP Option 42 to override If DHCP Option 42 is enabled, the NTP server can be changed. Enabled by default NTP server Time Zone Selects the time zone where the phone is located and control the date/time display...
Security Settings Web/SSH Access Web Access Mode Sets web page access protocol, the options are: HTTP, HTTPS, Disabled HTTP Web Port De nes the HTTP Web Port, defaut is 80 HTTPS Web Port De nes the HTTPS Web Port, defaut is 443 Web Session Timeout De nes the web session timeout, default value is 10 Minutes, the valid range is 1-60 minutes Web Access Attempt Limit...
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Example:192.168.5.222 192.168.5.223 192.168.7.0/24 Note: If both blacklist and whitelist are not empty, the blacklist is processed rst, followed by the whitelist. If WAN Side Web/SSH Access is set to Yes or Auto. Users can con gure the black List for WAN Side to ban WAN side web access.
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This feature allows users to validate server certi cates with our trusted list of TLS connections. The device Validate Server Certi cates needs to reboot after changing the setting. Default is enabled. This feature allows customer to choose desired Minimum TLS Version. Choices are: Unlimited Minimum TLS Version...
Con gure the RADIUS VSA provider ID to match the RADIUS server's con guration. The default value for RADIUS VSA Vendor ID Grandstream Networks Inc is 42397 Con gure the RADIUS VSA access level attributes to match the con guration of the RADIUS server. Incorrect RADIUS VSA Access Level Attribute settings will cause Radius authentication to fail.
● Both, prefer IPv6: Enable both IPv4 and IPv6 and prefer IPv6. Note: Make sure to reboot the HT841/HT881 unit for the changes to take effect. Allows users to con gure the appropriate network settings on the HT841/HT881 to obtain IPv4 address. Users IPv4 address type could select DHCP, Static IP or PPPoE.
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Speci es preferred DNS server to use when DHCP or PPPoE are set. Preferred DNS server 1-4 You can set up yo 4 Preferred DNS Servers. Con gures the IPv6 address of the phone: ● DHCP: all the eld values for the static IP mode are not used (even though they are still saved in the ash memory.) The FXO Gateway acquires its IP address from the rst DHCP server it discovers from the LAN it is connected.
SNMP Settings Enables the Simple Network Management Protocol. Enable SNMP Default is "No" SNMP Version Choose between (Version 1, Version 2c, or Version 3). SNMP Port Listening Port of SNMP daemon (Default 161). IP address of trap destination. Up to 3 trap destinations are supported. Users should enter the IP addresses SNMP Trap IP Address separated with comma (,).
The default mode is Bridge mode. Save the setting and reboot prior to con guring the HT841/HT881. When set to Yes, the HT841/HT881 acts as a UPnP gateway for your UPnP-enabled applications. UPnP = Enable UPnP support Universal Plug and Play. The default is No.
Note: Voice or RTP stream will never be limited. This function forwards all WAN IP tra c to a speci c IP address if no matching port is used by HT841/HT881 DMZ IP or in the de ned port forwarding.
Disabled by Default. de nes the allowed methods (protocols) and permissions for remote access and con guration of the device, two options can be selected ● Management Interface Only: allows remote access and con guration solely through the dedicated Management Access management interface ●...
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Enable OpenVPN LZO Compression Enable OpenVPN® LZO Compression. Default is Yes. OpenVPN Encryption Select the OpenVPN® Encryption. Default is BF-CBC 128 bit (default key). OpenVPN Diges Select the OpenVPN® Digest. Default is SHA1. OpenVPN Username Con gure the OpenVPN® username. OpenVPN Password Con gure the OpenVPN®...
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Checks if rmware le is with matching post x before downloading it. Firmware File Post x This eld enables user to store different versions of rmware les in one directory on the rmware server. Con g File Upload Con guration Uploads the con guration le in .txt or .xml formats Restore From Backup Con guration Restore From Backup Con guration...
Speci es when the rmware upgrade process will be initiated; there are 4 options: No: The HT8x1 will only do an upgrade once at boot up. Check every X minutes: User needs to specify a period in minutes. Automatic Upgrade Check every day: User needs to specify the start hour and the end hour of the day (0-23).
SIP protocol. An example of captured trace for the SIP header can be as shown below: ● HT841 --- 2024-05-03 04:43:09.827 SENDING TO 192.168.5.168:5060 REGISTER sip:192.168.5.168 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.66:5060;branch=z9hG4bK443123589;rport From: <sip:1000@192.168.5.168>;tag=1597031286...
Restore Factory Gives the administrator the option to restore the default con guration on the HT841/HT881. There are 3 types of factory reset: ● ISP Data Reset: All ISP (Internet Service Provider) con gurations that may affect the IP address will be Reset Type reset (including WAN static IP).
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Allows users to add the actual domain name, it will override the from header.This is an optional con guration. Allow DHCP Option 120 (override Con gures the HT841/HT881 to collect SIP server address from DHCP option 120. Default is No. SIP server)
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De nes NAT IP address used in SIP/SDP messages. It should only be used if required by ITSP. Proxy-Require Determines a SIP Extension to notify the SIP server that the HT841/HT881 is behind a NAT/Firewall. SIP Settings SIP Basic Settings Controls whether the HT841/HT881 needs to send REGISTER messages to the proxy server.
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Local SIP Port De nes local port to use by the HT841/HT881 for listening and transmitting SIP packets. Default value is 5060 De nes the local RTP-RTCP port pair the HT841/HT881 will listen and transmit. It is the HT841/HT881 RTP Local RTP Port port for channel 0.
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De nes T1 timeout value. It is an estimate of the round-trip time between the client and server transactions. For example, HT841/HT881 will attempt to send a request to a SIP server. The time it takes between sending SIP T1 Timeout out the request to the point of getting a response is the SIP T1 timer.
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The authentication in this case can be either with: The admin password if no SIP account is con gured on the HT. With the credentials of the SIP account if con gured on the ATA. This feature allows users to con gure SIP User Agent. If not con gured, device will use the default User Agent SIP User-Agent header.
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Check SIP User ID for incoming Checks the SIP User ID in the Request URI of the incoming INVITE; if it doesn’t match the HT841/HT881 SIP INVITE User ID, the call will be rejected. Direct IP calling will also be disabled. The default is No.
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RFC2833 End Events Count This feature allows users to customize the count of RFC2833 end events. Default is 3. Preferred Vocoder Con gures vocoders in a preference list (up to 8 preferred vocoders) that will be included with same order in Preferred Vocoder (in listed order) SDP message.
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HT841/HT881 support a range from 40 to 2000 ms. Default values are 300 minimum and 1100 maximum. Speci es the on-hook time for an on-hook event to be validated. HT841/HT881 support a range from 40 to On Hook Timing 2000 ms. Default value is 400.
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OnHook DC Feed Current This feature is used to adjust DC feed current. Call Settings Sets the delay time before automatic dialing begins after going off-hook on an FXO port. Offhook Auto-Dial Delay Range is 0-60 seconds, default is 0 No Key Entry Timeout Initiates the call within this time interval if no additional key entry during dialing stage.
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RFC3261 (a line) will play the hold music to the other side. Disable Call-Waiting Disables receiving a second incoming call when the line is engaged. Default is No. Disable Call-Waiting Caller ID Disables displaying caller ID when receiving a second incoming call. Default is No. Disables playing call waiting tone during active call when receiving a second incoming call.
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Sets “From”, “Privacy” and “P_Asserted_Identity” headers in outgoing INVITE message to “anonymous”, Send Anonymous blocking caller ID. Default is No. Anonymous Call Rejection Rejects incoming calls with anonymous caller ID with “486 Busy here” message. Default is No. Selects Soft switch vendors’ special requirements Examples of vendors: BroadSoft, CBCOM, RNK, Huawei, Special Feature China Mobile, ZTE IMS, PhonePower, TELKOM SA, Vonage, Metaswitch, CenturyLink, MTS, Oi_BR, Telefonica, GIBTELECOM.
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Enable CID per call Enable CID per call: Default is 82 Disable CID per call Disable CID per call: Default is 67 Direct IP Calling Feature Allow users to customize the Direct IP feature code. Default is Yes Direct IP Calling Direct IP Calling: Default is 47 CW Feature Allow users to customize the call waiting feature codes.
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Blind Transfer Feature Allow users to customize the Blind Transfer feature code. Default is Yes Enable Blind Transfer Enable Blind Transfer: Default is 87 Whether to disable the LEC (Line Echo Cancellation) function of the current call (only the current call is Disable LEC per call Feature prohibited) Disable LEC per call...
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(For example: sip.mycompany.com, IPv4:192.168.5.170, or IPv6: fe80::20b:82ff:fe75:211d). This is the primary SIP server used to send/receive SIP messages from/to HT841/HT881. De nes failover SIP server IP address (Supports both IPv4 and IPv6 addresses) or domain name provided by Failover SIP Server VoIP service provider.
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Use NAT IP De nes NAT IP address used in SIP/SDP messages. It should only be used if required by ITSP. Proxy-Require Determines a SIP Extension to notify the SIP server that the HT841/HT881 is behind a NAT/Firewall. SIP Settings...
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Local SIP Port while the Port Id is de ned as follows: FXS = 0, FXO1-8 = 1 – 8 for HT881 and FXO1-4 = 1 – 4 for HT841, in this case, if we set FXO pro le 1 local SIP port to 6060, then FXO1 will use port 6062, FXO2 will use 6064, FXO3 will use 6066 and so on…...
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De nes T1 timeout value. It is an estimate of the round-trip time between the client and server transactions. For example, HT841/HT881 will attempt to send a request to a SIP server. The time it takes between sending SIP T1 Timeout out the request to the point of getting a response is the SIP T1 timer.
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and doubling of T1 continues until it reaches the T2 value. The default is 4 seconds. SIP Timer D Con gures SIP Timer D de ned in RFC3261. 0 – 64 seconds. Default is 0. Replaces # by %23 in some special situations. Do Not Escape '#' as %23 in SIP URI Default is No.
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Checks the authenticity and integrity of incoming SIP messages for enhanced security. Check SIP User ID for incoming Checks the SIP User ID in the Request URI of the incoming INVITE; if it doesn’t match the HT841/HT881 SIP INVITE User ID, the call will be rejected. Direct IP calling will also be disabled. The default is No.
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Adjusts the transmit gain for inband Dual-Tone Multi-Frequency (DTMF) tones during call signaling. Inband DTMF Tx Gain The Valid range is -12-12 db, default is 0 Allows users to con g the DSP DTMF Detector Duration Threshold. DSP DTMF Detector Duration The default Duration is 30 ms.
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De nes the time period when the cradle is pressed (Hook Flash) to simulate FLASH. To prevent unwanted activation of the Flash/Hold and automatic phone ring-back, adjust this time value. Hook Flash Timing HT841/HT881 support a range from 40 to 2000 ms. Default values are 300 minimum and 1100 maximum. Adjusts the voice path volume.
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Disables the LEC per call base. Recommended for Fax/Data calls. Disable Line Echo Canceller (LEC) Default is No. Customizes ring frequency. Ring Frequency Valid options: 20Hz – 25Hz. Default is 20Hz. FSK Caller ID Minimum RX Level Sets the minimum signal level required to detect and receive FSK Caller ID information accurately. (dB) The valid range is -96 - 0dB.
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user dial more than just 3 digits if needed. Originally the phone will dial out immediately after dialing the third digit. Example 1: {[369]11 | 1617xxxxxxx} – Allow 311, 611, 911, and any 10-digit numbers of leading digits 1617 Example 2: {^1900x+ | <=1617>xxxxxxx} – Block any number with leading digits 1900 and add pre x 1617 for any dialed 7-digit numbers Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} –...
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This value should be used in case the PSTN provider uses line power drop to indicate call completion to the Enable Current Disconnect end point. In this case the HT841/HT881 will search for a power drop. This is a precon gured value of duration for a line power drop used by speci c service providers. For example,...
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Stage Method (1/2) This con guration is applicable for VoIP to PSTN calls and indicates one or two stage dialing methods. The time to wait before HT841/HT881 initiates the call via PSTN line. Min Delay Before Dial PSTN Number Default 500ms, range is from 50 to 65000ms.
Con gures a User ID or extension number that is automatically dialed when off-hook. Only the user part of a Offhook Auto-Dial SIP address needs is entered here. The HT841/HT881 will automatically append the “@” and the host portion of the corresponding SIP address.
● Sip Destination Port: Displays the destination port IMPORTANT WEB CONFIGURATIONS The HT841/HT881 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow a user to configure the HT841/HT881 through a web browser such as Google Chrome, Mozilla Firefox and Microsoft’s IE.
After first time log in attempt, the user will be forced to change his initial admin password. Changing User Level Password 1. Access your HT841/HT881 web UI by entering its IP address in your favorite browser. 2. Enter your admin password (default: The password found on the sticker on the back of the unit).
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Select voice menu option 01 to allow the HT841/HT881 to use DHCP. STATIC IP MODE Select voice menu option 01 to allow the HT841/HT881 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.
Select voice menu option 12 to enable/disable web access from WAN port. Press 9 in this menu to toggle between enable / disable. Register a SIP Account The HT841 supports 1 SIP account for FXS port, and 4 accounts for FXO ports, While the HT881 supports 1 SIP account for FXS port, and 8 accounts for FXO ports.
UPGRADING AND PROVISIONING The HT841/HT881 can be upgraded via FTP/FTPS/TFTP/HTTP/HTTPS by configuring the URL/IP Address for the FTP/FTPS/TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid URL for TFTP or FTP/FTPS or HTTP/HTTPS (default is HTTPS); the server name can be FQDN or IP address.
TFTP , FTP/FTPS or HTTP/HTTPS. 5. Make sure to check “Always Check for New Firmware”. 6. Update the change by clicking the “ Apply” button at the bottom of the page. Then “Reboot” or power cycle the HT841/HT881 to update the new firmware.
>Web/SSH Access page->Admin Password. For a detailed parameter list, please refer to the corresponding firmware release configuration template. When the HT841/HT881 boots up or reboots, it will send a request to download the xml file named “cfgxxxxxxxxxxxx.xml” followed by the binary file named “cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx”...
For example: if the MAC address is 000b8200e395, it should be keyed in as “0002228200333395” Reset from Web Interface (Reset Type) 1. Access your HT841/HT881 UI by entering its IP address in your favorite browser. 2. Enter your admin password (default: Found on a sticker on the back of the unit).
Factory Reset will be disabled if the “Lock keypad update” is set to “Yes”. If the HT841/HT881 was previously locked by your local service provider, pressing the RESET button will only restart the unit. The device will not return to factory default settings.
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