Grandstream 2-FXS Port Analog Telephone Adapter (HT802)

Works really well with VoIP.ms. Fairly easy to set up, you can use the local web config page or do the full blown device management that Grandstream offers for free. VoIP.ms has setup guides.

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.

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.

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!

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!

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.

These units work very well. There are a couple parameters that have to be changed. The instructions are straight forward and tech support is very helpful. Be sure to hit save once you change the communications parameters under each tab. If you click on a different tab without saving your changes, they will be lost. If you are comfortable with setup software for devices, it should take you about 20 minutes. Have tech support at the VOIP provider send you the field names to change and the values you should input or select. I found it easier to use the phone you connect it to in order to get the IP address, then type the IP address into your computer browser. This unit must be plugged into the same switch as your computer.