



📞 Stay connected, secure, and ahead with Grandstream HT-812 — your analog gateway to the future!
The Grandstream HT-812 2FXS Gateway is a professional-grade VoIP adapter designed to bridge traditional analog phones with modern IP networks. Featuring 2 FXS ports, dual Gigabit Ethernet ports, and a built-in NAT router capable of routing speeds up to 100Mbps, it delivers exceptional voice quality and network performance. Enhanced with TLS and SRTP encryption, it ensures secure communications, making it ideal for home and office environments transitioning to VoIP technology.
| ASIN | B01M2ZJQAF |
| Best Sellers Rank | 84,890 in Computers & Accessories ( See Top 100 in Computers & Accessories ) 637 in Routers |
| Guaranteed software updates until | unknown |
| Item model number | HT-812 |
| Language | Dutch |
| Manufacturer | Grandstream |
| Product Dimensions | 22.86 x 33.02 x 7.11 cm; 290 g |
T**E
Compatible with Zen's UK "BT digital voice" switchover
Bought as I'm a zen.co.uk customer, and have UniFi throughout the house (I don't use the Fritzbox). With the BT/UK "digital voice" switchover in full-swing, I wanted a means to keep our old land line alive. This worked perfectly. Configuration hasn't been utterly straight-forward, but I've finally found the following settings work (for Zen customers): - Ensure your NAT/firewall is forwarding the required ports through to the Grandstream (5060/UDP from 212.23.7.228, and 20000-30000/UDP from 62.3.88.0/28 and 62.3.88.16/28) - Primary SIP server: voip.zen.co.uk - Secondary SIP server: voip2.zen.co.uk - Prefer Primary SIP server: No - Outbound proxy: Blank/None - Allow DHCP Option 120: No - SIP Transport: UDP - NAT traversal: No - DNS Mode: A Record - DNS SRV: No - Fallback Timer: 60 - Use Request Routing ID in SIP INVITE Header: No - Sip Registration: Yes - Outgoing Call without Registration: Yes - Register Expiration: 5 - Enable SIP OPTIONS/NOTIFY keep alive: No - Local SIP port: 5060 - Local RTP port: 5004 - Random SIP port: No - Random RTP port: No - Enable RTCP: Yes - Hold Target Before Refer: Yes - Refer-To Use Target: No - Remove OBP: No - Support SIP Instance ID: Yes - Validate Incoming SIP Message: No - Use Privacy Header: No - Use P-Preferred-Identity Header: Yes - Use P-Access-Network-Info Header: Yes - Use P-Emergency-Info Header: Yes - Use P-Asserted-Identity Header: No - SIP REGISTER Contact Header Uses: LAN - Caller ID Fetch Order: Auto - DTMF: RFC2833 - SLIC: UK - Caller ID Scheme: SIN 227 - Dial Plan: (You may want to customise) ... { 999 | 10[015] | 11[129] | 11[68]xx | 1[45]7[1-2] | <0=44>8001111 | <0=44>845464x | <0=44>[1235789]xxxxxxxxx | <=441638>23456789]xxxxx | 1410[1235789]xxxxxxxxx | x+ }
A**N
Setting up takes some work. Call quality is fine.
Bought this HT512 to replace a Linksys pap2t which had died. The Grandstream shipped from USA and arrived with a USA power supply so first thing needed was a UK to USA mains adapter. Also included was a network cable. There was no user guide included (but this can be downloaded from the Grandstream website) and no phone tails to convert RJ11 to UK telephone socket. Luckily I could use the old Linksys tails. The HT512 has a WAN port and a LAN port which do different things. The WAN port is what you normally expect and is used to connect to an existing LAN (like on the |Linksys). The LAN port is a built in router/dhcp server etc. I was replacing an existing device so the WAN port is what I needed (not LAN). I used the web interface to set up the HT512. For setting up the 2 telephones, you need to get the settings from your SIP provider. Mine (Sipgate) had generic instructions for the Grandstream HT series. These showed all the settings on a single config screen, however this HT512 had the settings spread over 3 different screens. However, after finding where everything should go, I plugged the phone in and fired it it with a test call. It worked first time. In summary, the HS512 had a few shortcomings (no manual, USA power supply, no UK tails) and required a fair bit of detective work to set up but as a replacement for the Linksys pap2t, it is ideal. Even if the built in router/dchp server (LAN port) is not required.
C**S
Provided introduction to VOIP
The imminent arrival of FTTP prompted the investigation of VOIP to allow the ditching of the increasingly expensive landline which is now only used by ageing relatives and neighbours who have not moved to mobile. Since I do not intend to make outgoing calls, Sipgate provides a low cost (free), local number to which I can later migrate my BT number if it works well (at additional cost). The HT-812 allows the use of my existing analogue phones to assess the performance/reliability of VOIP as an alternative to the existing copper landline so I can use the money saved to pay for fast FTTP. Configuration is somewhat daunting due to the huge number of settings in the configuration GUI but only a few need changing. After updating the firmware to the latest version (see online manual) there are even more options available but most VOIP providers do supply settings for this series of adapters as well as the SIP codes, passwords etc. I also made these additional changes to make my phones sound like BT phones rater than US ones. In Advance Settings:- System ring cadence: c=400/200-400/2000; Dial tone: f1=350@-19,f2=450@-22,c=0/0; You will need one (or two) RJ11 (male) to BT (female) adapters in order to plug in your existing phones. For traditional phones these need to be masters including the ring capacitor. For modern DECT base stations the capacitor does not seem to be required. For my experiment I had to resort to adding some old ADSL filters which contain the ring capacitor, to my adapter cables. So to the results. The HT812 works perfectly. Audio quality is better than the landline but otherwise I can see no difference. I will have to see what the reliability of the VOIP service is but for the very low number of calls it will probably not matter. I will be migrating my BT number and then will enjoy not having to pay the line rental costs which will soon recoup the cost of the HT812! Do remember that migrating your number will cease the existing copper line and all services including ADSL/FTTC. If this appears too complex a process then it might be better to pay for Digital Voice (the same thing but configured by your ISP), but for my use case the VOIP gateway is an attractive option.
L**X
We switched from Ooma to a SIP provider (VOIP.ms, but there are lots of others). We got a few more features and are saving a little money. This was a little tricky to setup (it has lots of settings). Most of the settings are fine if left at default, but it's capable of encrypting traffic and a whole bunch of other stuff and there are settings for every possible feature. In fact, there are five or six options to backup/restore the configuration (seems like any of them will work, though).
L**S
Lo probé con Megacable conectando el módem a modo puente con este dispositivo usando la VLAN 559 y funciona de maravilla. Es importante que la cuenta SIP se puede tener activa en un dispositivo a la vez.
M**K
Just been fitted. Very happy. Great idea if you don't want a normal house telephone line. Works great with Starlink.
A**N
Just works. Purchased to replace a PAP2T that wouldn't ring the bell on a rotary phone.
G**N
Das Gerät an sich ist gut verarbeitet. Leider fehlt eine vernünftige Bedienungsanleitung, daher auch nur 2 Sterne für die Installation. Doch dank der genauen Beschreibung von Herrn Müller, konnte auch ich als Leihe das Gerät installieren und programmieren. Jetzt verrichtet es seinen Dienst. Es folgt auf ein DECT Gerät der Firma Wantec, was von der Installation her wesentlich einfacher gestaltet war, da es über Funk arbeitet. Allerdings war dieses Gerät mal wieder defekt (in 5 Jahren das 4) und auch nicht mehr zeitnah lieferbar. Der Grandstream kam schon einen Tag früher als angekündigt und ist auch recht ansehnlich. Über die Haltbarkeit kann ich natürlich noch nicht wirklich Auskunft geben, da er erst seit heute in Betrieb ist. Ich kann jedem nur empfehlen die Anleitung zur Einrichtung von Herrn Müller zu nutzen, dann funktioniert auch alles.....
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