Grandstream HT801 Single-Port Analog Telephone Adapter

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

ممتاز ،، ربطت تلفون عادي مع بدالة 3cx متوافق تماما

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.

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