Grandstream 2-FXS Port Analog Telephone Adapter (HT802)

I had a Cisco SPA112 analog telephone adapter (ATA) with Vonage for many years. People on the other end started hearing themselves as an echo. After many phone calls to Vonage customer service and 4-5 hours wasted talking to them, I decided to purchase the Grandstream HT802. I called Vonage and had them remote into my laptop to configure the HT802 to work with their network. After a reboot of the device it's been working fine ever since with no more echo issues. You can also configure the HT802 to make "offline" calls between two landline telephones - no phone service/network required. This could be used like an intercom between two phones in the same house or building or between a gate and a house/building. To see how to do this, search for "Make Offline Calls Between Landline Telephones - no Phone Network" on YT.

My VOIP provider only charges eighty-five cents per month for a phone number, and two cents per minute for calls anywhere in the US. But to get that deal, you need this device to allow a normal landline phone to work. This device plugs in to your router and can give you up to two analog phone lines with a normal dial tone on each line. Just plug it in to any phone outlet in your house after setup, and all phone outlets in your house will have one or two active phone lines. The setup with your VOIP provider and configuration of this device is a bit complex and might not work for those who are not familiar enough with networking. I had to have my internet service provider tell me how to open a port on the cable modem/router to get it to work. There is also a learning curve if you're not experienced at networking and setting up a SIP service. But with some patience, the setup process is not too bad. Tech support from your SIP provider should be available to you for free. I could do it quickly and easily again, now that I know how. Once it's working, you never need to do anything technical to it again. It just keeps working after the initial setup.

This is a good product and value. I have two of them and have not had any problems. I have tested with multiple VOIP ptoviders and the sound is fine. It looks cool and has taken a couple of falls without damage (magnet mount to wall). I have two and durability hasn't been an issue. The UI is a bit dated but works fine. It times out too quickly but has dozens of options. I use two for SOHO but the number of options could lean towards enterprise. I have not had anyone complain of lack of clarity.

Leaving a review because none of them seemed to mention that this works great with FreePBX. I am in no way a VOIP/PBX expert so I struggled to find instructions on how to get them working together. I also discovered that there must be a password length limitation on the Grandstream as it failed to register until I set a shorter device password/secret (FreePBX logs showed failed to authenticate). To get this working with FreePBX In FreePBX: 1) Applications > Extensions 2) Add Extension > Add New SIP (chan_pjsip) 3) Pick your extension number and set a Secret that is less than 14 characters. 4) Take note of the banner at the top indicating what Port your PJSIP is listening on (default 5060 UDP) Log into the Grandstream Web UI using the device IP address (default login is admin/admin) 5) Click FXS Port 1 or FXS Port 2 at the top 6) For Primary SIP Server, enter your FreePBX IP address AND the port (192.168.1.10:5060) 7) Scroll down to SIP User ID and Authenticate ID, enter the extension number from step 3 above 8) For Authenticate Password, enter the Secret you set for the extension. 9) I also set my Primary DNS IP (probably not necessary if you use DHCP) 10) Scroll to the bottom and click Apply. After that, on the Grandstream Status page you should see your FXS port as Registered after some time. You might need to reboot the device. Obviously repeat all steps for the other FXS port. If you google the Granstream model number their website has a PDF document that explains what all the options are within the configuration.

I bought one and it was defective, and the callcentric support team could not get it to work, so I sent it back. They replaced it with another one, and that one was worse then the first one... I decided to go with the Cisco SPA-192 instead, it may cost more, but I have wasted many hours and many days shipping time on these grandstreams, only to still be using my old device... Edit: The Cisco SPA192 arrived, i set it up easily, and it worked right away with Callcentric service. if you are looking for a device that just works as it should go with the Cisco SPA192...

Well, all good things come to an end, and the writing is on the wall with the Obi202 reaching end of life in December 2023. Rather than wait for things to just suddenly stop working someday, I decided to sign up for outgoing service through CallCentric and use this officially supported box. I'm glad I did. I have used my Obi202 and Google Voice for what feels like 15 years now. Free phone service has been great, except for $2/month for an incoming number through CallCentric that gave me e911 service and caller ID. However, people would often complain that they couldn't hear me that well over the past 2 years. I decided, with the announcement of Obi202's retirement, that I would sign up for CallCentric's outgoing plan and pair it with this box. Using the instructions on CC's site, I configured this box as instructed and immediately had service within minutes. I made a few test calls, and everything worked great, including my outgoing caller ID showing up as my Google Voice number, which CallCentric can do for you. This box helps me get much better sound quality than I had with the Obi202 + Google Voice combination. People on the other end of the phone say they notice a big difference when talking to me now, which is great. I don't mind paying a few bucks more every month for reliable, quality phone service at home. This box was a great deal under $40 and I will recommend it to friends and family.

Concurrent direct du SPA112 de Cisco. • Fonctionne beaucoup mieux que le SPA (j'en avais un pendant 2ans, je l'ai détruit à la masse...) • Ses connexions sont stables et rapides. • J'apprécie quelques options fondamentales en plus comme par exemple le fait de pouvoir couper l'appel sortant au bout d'un certain temps pour éviter des hors forfaits. • Il ne redémarre pas non plus à chaque fois que l'on fait une modification. • L'interface est plus simpliste que le SPA, ce qui peut rebuter. Ne pas tenir compte de cela. C'est exclusif (ça mérite un vote utile ☺), je vous donne les points clé de config de ligne pour le faire marcher. (et si ça marche pas vous n'êtes pas loin). Cette config est pour un téléphone. Account Active: Yes Primary SIP Server: SIP User ID: Authenticate ID: Authenticate Password: Outgoing Call without Registration: No Dial Plan: {[x*]+} (info: ce paramètre est en cause si réceptionne call mais compose et n'émet pas) SRTP Mode: Disabled SLIC Setting: European CTR21 Caller ID Scheme: ETSI FSK prior ringing with (LR+)DTAS Outgoing Call Duration Limit: 59 (évite surfacturation >1h, dépend de l'opérateur)

設定は、インターネットにあった先人達の情報をみてやりました。想定通りつながりました。 これと、wifi受信機を複合機をセットで組み、どこにでも据え付けできるようになりました。もっと早く買えば良かったです。

Por fin he encontrado un dispositivo que permite usar el teléfono (encapsulado en VLAN) en una conexión de fibra. La interfaz web es un poco lenta, pero tiene múltiples opciones y se pueden introducir las credenciales SIP sin problema. Muy recomendable.

Grandstream GS-HT802 was delivered today, the seller is ThePhoneGuy-Canada. I mention the seller because technical issues and technical support issues, as seen in this comment section, seem to be seller-specific. In my case, the first issue came immediately, HT-802 web interface did not want to accept user name admin, password admin -- these default names are stated in every Manual available in the Internet and on the Manufacturer site. I set a ticket at the Grandstream tech.support and get a response within 30 minutes - "Look at the bottom of the device", this solved the problem. Again, based on comments in this ection and elsewhere, I expected no response for days or empty response -" go for support to the seller". After the issue was solved, a configuration process went smoothly - I used a Wiki at VOIP.ms, which is my current provider. Within 30 minutes I was able to make calls, the quality of calls was better compared to my conventional land line. Summary: - tech.support was very fast - sophisticated configuration was smooth (use proper help!) - initial impression about the quality of calls: great! Obviously, the reliability of the device is very important, so, I will come and update this review "if/when"

brought this to connect SIP to my analog pbx works without any issues