Grandstream HandyTone HT801 ATA

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.

Arrives in a simple box with absolutely no instructions, a PSU and a short Ethernet lead. A really annoying thing is that they don't tell you that the default password is admin. Easy to connect, but the setup is very complex, you need instructions from your supplier. To connect, you plug an Ethernet cable in to the port on the Grandstream, and the other end in to your router. You then need to find the Grandstream, so you can access the web configuration pages it has. Your router assigns an IP to the Grandstream, and there is no easy way to find what that is. Fortunately, my router displayed it. I then went to the Grandstream and changed the IP to a fixed address, so I could always find it. I believe there is a discovery tool you can download to find it. Your existing telephones plug in to the Grandstream, although you will probably need a UK to US adapter. These are cheap, costing as little as £2. You can plug in old phones in to the Grandstream, or modern phones. Old phones, that use pulse dialling, require a converter that is unfortunately expensive. You can find these on eBay. One will cover you for several phones. You can mix modern and old. I have two old phones and one dect unit on the same Grandstream. To replicate the old BT ring "dring, dring" you need to carefully set up the Grandstream. In advanced settings, you need to set the Ring Cadence to c=400/200-400/2000 To get this all to work, you need to sign up with a SIP provider. There are plenty around. Most of them will detail the setup of the Grandstream you need to access their service. They will provide you with a telephone number. Bought this on recommendation because I was having problems with a Cisco 191. Not only is the Grandstream much cheaper, it works perfectly. Including DTMF, which is very dodgy with the Cisco. Would highly recommend.

This is a great ATA. Having the option to configure for pulse dial as a standard option is a real bonus. Easy to set up. Using with a BT746 phone and sipgate in the UK.

The unit comes without instructions, but they are available online. To set it up, I needed to combine the procedures in the online manual with instructions from my VOIP provider specific to this unit. This met with limited success initially, but to be fair it was the instructions from my provider that were the problem and a support call from their helpline resolved the issues. The next problem was that my real vintage phone unit wasn't compatible and did not ring on incoming calls. Again, this is more likely an issue with the old handset and a new analogue model works fine. For UK customers, you will also need an adaptor plug for female UK to male US. Overall, this works well and I now have a low cost VOIP landline. Recommended, but be prepared for the setup.

Bought this when I moved broadband provider to Open Infra who do not offer a phone. Using Andrews & Arnold's VoIP service, its easy to set this up following the instructions on their support site. As for the comments about the US style plug in the box, it uses micro USB so any USB charger with appropriate cable will do.

Bought this to allow an elderly relative living in a annex to use a phone in the way she was accustomed without paying extortionate BT charges to keep the old copper service live (have fttp installed). Unfortunately unit won't ring the old handset and despite many attempts to get this working (grandstream support were responsive but ultimately ineffectual) I gave up, returned unit and bought a more expensive cisco unit which just worked (tm). If you are prepared to spend hours setting arcane settings in a rather clunky UI (and updating the firmware seemingly with every support response) then you may well have better luck than me - the unit certainly did the sip connection part but this really felt like a throw-back to 1990's electronics - and I no longer have the patience for that....

Compact design, not easy to configure but got lots of help from my selected VoIP service provider, A&A. The unit is supplied with an American power supply but that isn’t needed, the power can be provided by any usb outlet using a micro usb lead. You will need a RJ11 plug to BT socket adaptor. There are no instructions supplied but can they can be found on line.

No instructions are provided in the box so configuration is dependent on third-party websites. The set-up screens (accessed via IP address) are long and require a large quantity of data to be entered manually. This makes it really difficult to know what’s wrong if it doesn’t work first time. There are numerous passwords required and it’s unclear which ones do what. The set-up tool does nothing to acknowledge password entry, leaving the fields blank for “security purposes”. So, you don’t know if a failure is due to an incorrect or missing password. Andrews & Arnold were helpful and responsive to help get me set-up in the end. But, their online instructions lacked clarity too. A redesign of the set-up screens with clear step by step instructions are required. Once set-up, the device works well and is OK value for money. So 3 stars from me overall.

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

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.

Hace su trabajo perfectamente una vez configurado. Es un poco entretenido para configurarlo correctamente. Yo lo logré después de ver un vídeo de naseros.com. Desde aquí, Muchas gracias a Naseros por sus vídeos para configurar distintos Routers y auxiliares de telefonía.

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