Grandstream Networks HT841 Manuale utente

Grandstream Networks, Inc.

HT841/HT881

User Manual

HT841/HT881 – User Manual

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. This series of enterprise analog FXO gateway can convert SIP/RTP IP calls

into conventional PSTN calls, and it is available in two models, the HT841 and HT881, with 4 and 8 FXO ports, respectively.

Both models share the same installation process.

This User Manual aims to guide you in effectively operating and managing your HT841/HT881 Analog FXO Gateway,

empowering you to utilize its numerous enhanced features, including seamless and swift installation.

PRODUCT OVERVIEW

Feature Highlights

The following table contains the major features of the HT841/HT881:

HT841/HT881 Feature Highlights

HT841/HT881 Technical Specifications

The following table resumes all the technical specifications including the protocols/standards supported, voice codecs,

telephony features, languages, and upgrade/provisioning settings for the HT841/HT881.

HT841/HT881

●2 FXO profiles supporting 2 different SIP Servers.

●HT841: 4 FXO RJ11 Ports for PSTN Connections.

●HT881: 8 FXO RJ11 Ports for PSTN Connections.

●PoE in Support for NET2 port.

●2x 10/100 Mbps dual RJ45 network ports: NET2/NET1.

●Advanced security protection with SRTP.

●Wide range of caller ID formats.

●T.38 Fax for creating Fax-over-IP

●Strong AES encryption with security certificate per unit

●Failover SIP server automatically switches to secondary server if main server loses

connection.

●Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-

based environments.

●Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning

●Automated provisioning options include TR-069 and XML config.

●Lifeline support (FXS port will be hard-relayed to FXO port) in case of power outage.

Interfaces

Telephone Interfaces

One (1) RJ11 FXS port.

HT841: Four (4) RJ11 FXO PSTN line port with lifeline support.

HT881: Eight (8) RJ11 FXO PSTN line port with lifeline support.

Network Interface Two (2) 10/100 Mbps ports (RJ45) with integrated NAT router

LED Indicators POWER, FXO, FXS, NET1, NET2.

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.723.1, G.729A/B, G.726, 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 3 REN: Up to 1km on 24 AWG

Caller ID Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID

Disconnect Methods Busy Tone, Polarity Reversal/Wink, Loop Current

Signaling

Network Protocols TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, FTP/FTPS, ARP/RARP, ICMP, DNS, DDNS, DHCP,

NTP, TFTP, SSH, Telnet, STUN, SIP (RFC3261), SIP over TCP/TLS, SRTP, TR-069

QoS Layer 2 (802.1Q VLAN, SIP/RTP 802.1p) and Layer 3 (ToS, Diffserv, MPLS).

DTMF Methods In-audio, RFC2833 and/or SIP INFO

Provisioning and Control HTTP, HTTPS, SSH, FTP, FTPS, Telnet, SSH, TFTP, TR-069, secure and automated

provisioning using AES encryption, syslog

Security

Media SRTP

Control TLS/SIPS/HTTPS

Management Syslog support, SSH, Telnet remote management using web browser.

Physical

Universal Power Supply POE Input: 48 V / 0.5 A

DC Input: 12V/1A

Environmental

Operational: 32° – 104°F or 0º – 40ºC

Storage: 14° – 140°F or -10º – 60ºC

Humidity: 10 – 90% Non-condensing

Dimensions and Weight

●HT841/HT881

Dimensions: 190mm x 100mm x 28mm (L x W x D)

Weight: 0.46KG

HT841/HT881 Technical Specifications

GETTING STARTED

This chapter provides basic installation instructions including the list of the packaging contents and also information for

obtaining the best performance with the HT841/HT881.

Equipment Packaging

The HT841/HT881 FXO Gateway package contains:

HT841/HT881 Package Content

HT841/HT881 Ports Description

The following figure describes the different ports on the back panel of the HT841/HT881.

HT841/HT881 Ports Description

Compliance

Compliance FCC/CE/RCM/IC/UKCA

Check the package before installation. If you find anything missing, contact your system administrator

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.

HT841/HT881 Supported ports

Connecting HT841/HT881

The HT841/HT881 is designed for easy configuration and easy installation. To connect your HT841/HT881, please follow the

steps below:

Connecting HT841/HT881 using WAN

When connecting HT841/HT881 using the NET1 port, it will act as a simple DHCP Client.

1. Insert a standard RJ11 telephone cable into the FXS port and connect the other end of the telephone cable to a standard

touch-tone analog telephone.

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.

4. Power, NET1/NET2, FXO, and FXS LED will be solidly lit when the HT841/HT881 is ready for use.

Connecting HT841/HT881 using LAN

When connecting the HT841/HT881 using the NET2 port, it will act as a router and DHCP serving addresses, the devices

connected with HT841/HT881 LAN will pull DHCP addresses from your HT841/HT881.

1. Insert a standard RJ11 telephone cable into the FXS port and connect the other end of the telephone cable to a standard

touch-tone analog telephone.

2. Connect a computer or switch to the NET1/NET2 port of the HT841/HT881 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. If the PoE switch is used in step 2, this step could be skipped.

4. Power, NET1/NET2 and FXS, and FXO LED will be solidly lit when the HT841/HT881 is ready for use.

●Used to connect HT841/HT881 to your router or gateway using an Ethernet RJ45 cable.List

Item 2

POE IN PoE supported port.

FXS FXS port to connect analog phone / fax machine to HT841/HT881 using RJ11 telephone cable.

FXO 1...4

FXO 1....8

FXO ports to be connected to physical PSTN line from a traditional PSTN PBX or PSTN

Central Office.

HT841 supports 4 FXO ports and HT881 supports 8 FXO Ports.

Reset Factory reset button. Press for 7 seconds to reset to factory default settings. Quick press will

only reboot the unit.

Note

HT841/HT881 supports switching the working mode of NET1 and NET2 on the Web User interface.

The default Device Mode for the HT841/881 is bridge. NET1 and NET2 ports function almost identically in bridge mode.

Note

The default Device Mode for the HT841/881 is bridge. NET1 and NET2 ports function almost identically in bridge mode.

Connecting the HT841/HT881

HT841/HT881 LEDs Pattern

There are five (5) LED types that help you manage the status of your HT841/HT881.

HT841/HT881 LED Patterns

Please make sure to enable NAT Router under Web GUI 🡪 Basic Settings 🡪 NAT/DHCP Server Information & Configuration 🡪

Device Mode.

LED Lights Status

Power LED The Power LED lights up when the HT841/HT881 is powered on and it flashes when the

HT841/HT88 is booting up.

NET1 LED The NET 1 LED lights up when the HT841/HT881 is connected to your network through the

NET1 port.

NET2 LED The NET 2 LED lights up when the HT841/HT881 is connected to your network through the

NET2 port.

FXS LED

The FXS LED indicates the status of the FXS port phone on the back panel.

OFF – Unregistered

ON (Solid Green) – Registered and Available

Blinking every 500 ms – Off-Hook / Busy

Slow blinking – FXS LED indicates voicemail

HT841 HT881 LED Patterns

Application Description

IP PBX / SIP Server with HT841/HT881

A SIP proxy server such as UCM6xxx can be deployed with the HT841/HT881 series. In this environment, the SIP server handles

SIP registration and call control, and the HT841/HT881 processes media conversion between IP and PSTN calls.

There are 2 ways to configure HT841/HT881 when using with a SIP Server:

1. With SIP accounts configured on the Channels page. In this case, the HT841/HT881 act like an endpoint requesting

registration from the SIP 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.

2. Without SIP accounts. In this case, you simply have to configure the SIP Server to perform forwarding of the SIP INVITE

message with the FXO destination number to the gateway’s IP Address. The HT841/HT881 will receive the digits and

immediately forward them on the FXO lines to the destination PSTN. Most of the configuration on the Gateway for this

case will remain the default, except Stage Method needs to be set to 1, and SIP Server IP Address/DNS name has to be

filled.

Functional Diagram of IP-PBX & HT841/HT881

For incoming calls from the PSTN analog endpoints to the HT841/HT881, the device will auto-forward each call to a

configured IP extension. The SIP Server can then route the call based on its own configuration or IVR system.

FXO LED

The FXO LEDs indicate the status of the FXO ports-phone on the back panel.

OFF – Unregistered

ON (Solid Green) – Registered and Available

Blinking every 500 ms – Off-Hook / Busy

Slow blinking – FXO LEDs indicate voicemail

All LEDs display green when ON. The Ready light will only be ON when the network interface is ready and the Web User

Interface is accessible.

During a firmware upgrade or configuration download the following LED pattern will be observed: Power, Ready, and NET1

LEDs will be ON. The FXO port LED will keep flashing during download and then stay OFF while the new files are written. The

entire process may take between 20 to 30 minutes. The firmware upgrade is complete when you can login into the web

configuration pages.

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).

GXW42xx & HT8x1 Scenario/Toll-Free Calling between Locations

In this diagram, configure the SIP Server field to be the IP Address of the other gateway (i.e. configure the IP address of the

FXS gateway to be SIP Server of HT841/HT881 and vice versa). Please be sure you set SIP Registration to No.

Expected Call Flow: Analog Phone (GXW42xx) picks up and dials destination PSTN number. The call gets routed to the

HT841/HT881 which dials out the digit string onto the FXO Lines, thus reaching the destination PSTN endpoint. In reverse,

incoming calls from PSTN endpoints will be routed automatically to the FXS Gateway through HT841/HT881.

FXS and FXO Gateway Configuration Example

FXS/FXO Lifeline Implementation

The “Life Line” feature is a telecommunications feature that ensures users can still place or receive a Public Switched

Telephone Network (PSTN) call in emergency situations or when there is a disruption in the internet service (SIP registration

loss). The feature operates by connecting the PSTN line to an analog phone that is connected to the FXS port.

GXW42xx Gateway HT841/HT881 Gateway

Advanced Settings

STUN Server – Blank

Use Random Port – No

Advanced Settings

STUN Server – Blank

FXO lines

Wait for Dial Tone – Y or N (whichever works for your PSTN Service Provider)

Stage Method – 1

Unconditional Call Forward to VOIP:

ch1-8:444; @ch1-8:p1; ch1-8:5060++;

Channels

1-8 5060 Profile 1

Local SIP Listen port (For VOIP to PSTN calls) – 5060++

Profile 1

SIP Server – Set it to IP Address of GXW42xx

SIP Registration – No

NAT traversal – No

Profile 1

SIP Server – Set it to the IP Address of HT841/

HT881

SIP Registration – No

Outgoing Call without Registration – Yes

NAT traversal – No

FXO/FXS Lifeline Support

*Refer to the following lifeline port info table:

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). When this happens, the

PSTN line will be seamlessly connected to the analog phone connected to the FXS port. This allows users to make and

receive traditional phone calls through the PSTN network, providing a backup communication method during

emergencies or internet outages.

Always Connected Mode: In Always Connected mode, the Life Line feature is permanently active, meaning the PSTN line

is always connected to the phone connected to the FXS port. In this configuration, VoIP calls will not be allowed. This

mode ensures that the device prioritizes the traditional PSTN line for all communications and does not rely on VoIP

services.

Always Disconnected Mode: In Always Disconnected mode, the Life Line feature is disabled, and the device will only

allow users to make and receive VoIP calls on the FXS port. PSTN calls via the FXS port will not be possible in this

configuration. This mode is suitable for situations where the user wants to use VoIP exclusively and does not require the

fallback option of PSTN connectivity.

FXS Telephony Implementation

When the FXS port on an FXO gateway is connected to an analog phone, the following basic telephony operations can

typically be performed:

Placing a Phone Call

To make the outgoing calls using your HT841/HT881

Model Lifeline Port Mapping

HT841 FXS FXO4

HT881 FXS FXO8

1. Pick up the handset of the connected phone.

2. Dial the number directly and wait for 4 seconds (Default “No Key Entry Timeout”); or

3. Dial the number directly and press # (Use # as dial key” must be configured in web configuration).

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.

Direct IP Calls

Direct IP calling allows two parties, that is, an 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 HT841/HT881 and other VoIP Devices, have public IP addresses, or

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).

The HT841/HT881 supports two ways to make Direct IP Calling:

Using IVR

1. Pick up the analog phone then access the voice menu prompt by dialing “***”

2. Dial “47” to access the direct IP call menu

3. Enter the IP address after the dial tone and voice prompt “Direct IP Calling”

Using Star Code

1. Pick up the analog phone then dial “*47”

2. Enter the target IP address.

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.

Note: When completing a direct IP call, the “Use Random SIP/RTP Port” should be set to “NO”.

CALL FEATURES

The HT841/HT881 supports all the traditional and advanced telephony features, the following table summarizes each key with

its corresponding Call Feature.

No dial tone will be played between steps 1 and 2 and destination ports can be specified using “*” (encoding for “:”) followed by

the port number.

Key Call Features

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