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 great little device that does not try to dumb things down for users, as it offers a huge number of options. It works great with both of my deep discount VOIP providers, Anveo and Future-nine, thereby giving me "DIY" landlines for well under $100/year. Of course, if you want the reliability of a standard landline, you will need to put this device, your router, and your modem onto a UPS--but again that's a modest upfront cost compared to the monthly cost of most full-service analog landlines. There are a few drawbacks worth mentioning, however. One is that to update the firmware from Grandstream's web site, you must use an http, not an https URL. That seems a bit backwards for 2017, and suggests the company is not very security conscious. In fairness, the device offers complete control over the update URL, so you can also download firmware from a machine on your local network including over https. Obviously the device is targeted for enterprise users (which I am not), who will want to roll out configuration and updates from a central internal server. The second problem I had was that my phone would start randomly ringing or just emitting weird partial rings, with seemingly no caller. Upon looking at a packet trace from my border router, it turned out the Grandstream was being attacked remotely. Though the Grandstream is behind a NAT, I had configured the device to use UPnP, which made it open a port to the outside world on my router. I don't think any of the attacks were actually succeeding in doing anything, but it was annoying that they made my phone ring at odd hours. An easy solution was to configure the Grandstream to use keepalives instead of UPnP for NAT piercing. In fact, I'd initially used UPnP because I thought keepalives might not be reliable enough for incoming calls. However, the keepalives work great and prevent random outside IP addresses from messing with my Grandstream. Since keepalives are the easiest option to set up, this is really just a result of my overthinking things. But again, they don't inspire confidence in the device's security, so really only consider using this on a secure network.

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.

Do not use anything from Grandstream. This came with no user manual or instructions. I got Ring Central to help me and we got to a place where we needed a password which was supposed to be on the device. The # did not work. To get help, I had to jump through hoops and answer questions that were not relevant for hours. After five hours now, I have to wait another 24 hours for an answer! I tried Amazon and they also could not help me except to give me Grandstream's website. Horrible customer service!

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.

Background information: OBiTALK discontinued, on 6/7/2022, support for a variety of Obihai devices, including the OBi110 that we have been using as our VoIP ATA. As the result, we no longer had VOIP service from Anveo for our telephones and fax machine. We had been using our OBi110 for years, for both voice and fax communication, with our Anveo obitalk account. In particular, our fax machine worked fine with this set up. Based upon information from Anveo. we purchased the Grandstream HT-802 as a replacement for the Obi110. I was not able to obtain detailed open print or even sufficient) information anywhere 2 get the HT-802 to work with my Anveo account. It turned out that there was no way to get the HT-802 to work with my Anveo Obitalk account. With significant (and exceptionally good) assistance from the Anveo support rep, was able to get the HT-802 to function, however, it turns out that the HT-802 likely does not support fax communications, even at the reduced speed VoIP communications rate configuration of the fax machine, 9600 baud. By the way, Grandstream does not offer support for the HT-802. You may be able to obtain some support from the Grandstream online forum, however, if you do receive responses to your questions , the responses may be slow in coming. Outline of setup/configuration steps: Note: the following assumes that you have using Anveo as your VoIP service provider. If you have been using a different VOIP service provider, the following, in all likelihood, does not apply to you. 1. The first thing that you are going to need to do is open a support ticket with Anveo, as many of the steps that are required to convert from and Obitalk account using an OBi110 to “regular” Anveo account using a Grandstream HT- 802, require Anveo to implement. 2. Ask Anveo to convert your oi talk account to a “regular” “free” account. Ask them to credit the unused dollar amount portion of your obitalk account back to your account so that you can use for your “regular” account. 3. You may need to change your IVR/Call Flows to the default call flow. (My call flows that worked with obitalk/OBI110 no longer functioned correctly with the HT-802.) Anveo support can make this change for you. 3. Log into your account at Anveo: 3a. Note your Anveo account number. This is your SIP User ID 3b. In “SIP Registration Details: Note the part that follows the “@”of your SIP URL. Mine was “sip.anveo.com:5010 “ You will need this to configure the HT-802. 3c. In “SIP Registration Details: Note your password. You will need this to configure the HT-802. (The SIP password is assigned by Anveo and is not the same as the password to log in to your Anveo account. 3d. In the Call Security menu, increase “Block calls with call rate more than” to $0.04 4. The only screen that I had to configure at the webpage for my HT-802 was at the FXS1 tab. My configuration was as follows: Summary of FZS1 configuration for Anveo for the HT-802: Primary SIP Server: sip.anveo.com:5010 SIP User ID: your Anveo account number Authenticate ID: your Anveo account number Authenticate Password: your Anveo SIP password Name: Your name Maximum Number of SIP Request Retries:4 SIP T1 Timeout: 1 sec SIP T2 Interval: 4 sec 5. After you have configured the HT-802, the Status tab should show Port Status = “Registered” 6. After your HT-802 shows the status as registered, verify the following: 6a. Ability to make outgoing telephone calls. Note: You will need to add a one at the beginning of the telephone number that you are dialing. Example: 1-123-456-7890 6b. Ability to receive telephone calls. Note: the ringing that the calling party hears may have second ring delayed by a few seconds. This is normal. 6c. The following applies if you were paying Anveo for the optional E911 service. Make a test call to your local 911 and verify that they have the correct address for you. 7. For sending and receiving faxes, plan to use the functionality from your Anveo account as your fax machine will likely not work with the HT-802. Good luck!

Workes well

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