Grandstream HandyTone 801 Single-Port Analog Telephone Adapter (HT801), Corded, Black.

Works well with VOIP service from voip.ms (which is very inexpensive). It does require care in setting up.

Replaced my Nettalk device with this one and I couldn't be happier. Works flawlessly without issues and sound quality is great. Just the compact size to fit in our room. No drops due to device hangs up given a constant stable connection. Great deal for this device for VOIP calls. Still works over many months for durability.

I've started to collect analog phones but have never been able to actually try them out, so I decided to get an ATA and cheap VoIP service so I could give them a spin. The HT801 was recommended as a good ATA for analog telephony hobbyists/enthusiasts like me. There is no guidance on how to get started with the device in the box, but luckily I found a manual on their website (which interestingly says it should come with a quick start guide). There are a LOT of settings you can fiddle with on the web UI. One thing I did was I turned off SSH access which unexpectedly seems to also disable the web UI, so I'm glad my VoIP provider offers setup instructions specifically for the HT801 and I got it configured and working before discovering this... I'll have to reset it and reconfigure everything to get that web UI back. I also was able to turn on pulse dialing support, which is crucial for using my rotary phones. But yeah, after everything was set up, I was able to place and receive calls no problem with very good audio quality for phones at least 50 years old. The ringers on the phones also worked fine when receiving calls, which is something I was slightly worried about. Overall I'm pleased with this little device and excited to do more with it. It'd be 5 stars if not for the fact that now I'm going to have to reset it and start all over due to the unexpected "no ssh, no web UI" problem.

I had an Obi 100 for 10 years. All of a sudden, it stopped sending touchtone (DTMF) and I started having frequent issues where callers couldn't hear me. This device has pretty much the same feature set, and my touchtone works again and callers can hear me again. I set up weekly firmware updates and changed the admin password. All I needed to get going was my provider's URL and my account username and password (sip credentials, not my website credentials) to get going. There are a thousand other nerd knobs I could tweak, but so long as it's working, I'm not going to mess with it.

Good luck installing this device. it is NOT plug/play. I suspect I got a refurb/open box item because the packaging for the cables was already torn open and the instruction manual was missing. I found the manual online, but the instructions did NOT match the screens on the web app/portal so I could not figure out how to configure this. I assume a networking specialist would find this simple, but for anyone else, good luck.

As I tried to set this up, I could not get past the login screen because the admin password kept failing. I wrote a ticket to Grandstream support (had to create an account) and they responded the same day. There is a sticker on the back of ---at least my, device with the MAC address and the admin password. This is not documented anywhere. Another thing...they dont tell you what goes in the Username box. I was trying admin, Admin, administrator, Administrator. But...in the end it was just lowercase admin, using the password on the sticker. Hopefully this saves somebody some time. For now, I am giving the purchase 5 stars for quick the quick support, but I have not actually tested it yet (will update if I change my mind).

I bought to replace the MagicJack that I was not happy with. This is the V2 unit not the V1. Someone said the password didn't work as admin. This is true for the V2 the password is printed on the label on the back of the unit. Set up was not the same as the sample shown on the voip.ms web site, but it will set up and works great once set up. Make sure you update the firmware to latest from Grandstream web site. I couldn't be happier with HT801 V2 and voip.ms combo. Saving over $20/month over Spectrum. Once set up the quality of calls and the features are really good.

I have a small PC-based phone system for my house. It's cheaper than any kind of IP phone from any company, but I didn't want a Cisco desk phone wired in, so I bought this and connected it to my PBX and then plugged in my Panasonic 5 station cordless phone set. It was easy to set up, the instructions were clear and covered every case - and unlike many pieces of modern electronic, has never had to be reset in order to work.

Works perfectly converted digital line to analog for FAX connectivity. But it works for analog phone lines too.

The Grandstream HandyTone HT801 is a solid and dependable little device. Setup was quick and straightforward, and it worked flawlessly with my VoIP service right out of the box. The build quality feels good, and it’s small enough to fit neatly beside my router without taking up space. Audio quality is excellent — clear and consistent with no dropouts or echo. Configuration options are flexible for those who like to fine-tune settings, but it’s also simple enough for a basic plug-and-play setup. Overall, a great product that does exactly what it should. Perfect for bringing an analogue phone into a modern VoIP setup.

اذا تلفونك قديم او لاسلكي وتريد استخدامه كتلفون شبكة(ip phone, voip)، هذا هو الحل

Tired of paying 70 dollars a month for mostly people I don’t want to talk to to try to scam me, sell me some service I don’t need, etc, I looked at options. Keeping the number I’ve had for 30 years, and having a single number that mom and a few others could call and talk to whoever is home is nice. Giving this number to most businesses means that telemarketing calls don’t come to my cell phone. This was the same device used by a 3rd party provider called freephoneline.ca. I decided to try to purchase the device and pay a one time fee to connect with them vs pay a VERY small monthly fee. Everything is working great. Calls are reliable. Device works fine, and by unhooking the telco access at my inside junction, and connecting this to one jack, all of my existing jacks work. The device is great, but I would not recommend this (or any other similar device) unless you have a good support with your provider (I didn’t - but they had an online user forum and this was the device they use) or you are technically strong (30 years in IT). Many people would be better off getting the device with support, through their provider. Setup is easy, but there are 2 or 3 dozen settings many of which need to match your provider, or else you could have no calls, dropped calls, audio but no voice, or other strange things. I was able to get the correct settings online and had it set up and working in about 15 minutes. It has been working great for a couple of months now and I’m glad I did this.

Ho preso questo prodotto per utilizzarlo come adattatore VoIP con ftth tiscali, per fare a meno del modem tiscali: perfettamente funzionante. La configurazione richiede una certa familiarità con reti e voip, ma, sapendo dove mettere le mani, è semplice. Interfaccia web essenziale ma completa. Il consumo idle è di appena 1,6 watt. Per chi fosse interessato, ecco le parti più rilevanti della configurazione: # ADVANCED SETTINGS system ring cadence: c=2000/4000; Dial Tone: f1=425@-12,f2=425@-12,c=200/200-600/1000; Ringback Tone: f1=425@-20,c=1000/4000; Busy Tone: f1=425@-20,c=500/500; Reorder Tone: f1=425@-12,c=250/250; Confirmation Tone: f1=350@-11,f2=440@-11,c=100/100-100/100-100/100; Call Waiting Tone: f1=425@-12,f2=425@-12,f3=425@-12,c=400/100-250/100-150/14000; Prompt Tone: f1=350@-17,f2=440@-17,c=0/0; Conference Party Hangup Tone: f1=425@-15,c=600/600; Weak TLS = Enable Weak TLS Ciphers suites send SIP log = yes # FXS SETTINGS Account active = yes Primary SIP server = ims.tiscali.net Outbound Proxy = srvrm.p.ims.tiscali.net SIP transport = UDP SIP user ID = nnn Authenticate ID: [email protected] Autenticate password: ppp Name: nnn DNS Mode = SRV Tel URI = yes SIP Registration = yes unregister on reboot = yes Outgoing Call without Registration: no Register Expiration: 60 Reregister before Expiration: 1 Enable SIP Options/notify keep alive: OPTIONS Allow Incoming SIP Messages from SIP Proxy Only: yes Use P-Access-Network-Info Header: no [fondamentale, se no le chiamate in uscita vanno in errore] SIP REGISTER Contact Header Uses: wan address SIP Timer D: 50 DTMF Payload Type: 97 (modificare iLBC Payload Type) Preferred DTMF method: RFC2833 (1-3) SUBSCRIBE for MWI: no Enable 100rel: yes Preferred Vocoder: PCMA, G729 iLBC Payload Type: <> 97 altrimenti non può essere impostato DTMF Payload Type: 97 Fax Mode: pass-through SLIC Setting: European CTR21 Caller ID Scheme: ETSI-FSK during ringing configurare inoltro porte 5050 udp, 5004 udp sul firewall