Visit the Grandstream store
Grandstream HT802 Two-Port Analogue Telephone Adapter
TLS and SRTP encryption, 3-party conferencing and T.38 fax, compatible with Grandstream UXM IP PBXs.
- Encryption: TLS and SRTP
- 3-party conferencing
- T.38 fax
- Compatible with Grandstream UXM IP PBX
Ref.: HT802
519,00 DHincl. VAT
Only 2 left in stock
Earn 6 DH in loyalty credit with this purchaseHow does it work?Loyalty credit capped at 300 DH per order.
Receive it between Wed 23 Sept and Fri 25 Sept — delivery 40 DH incl. VAT all over Morocco
Free in-store collection: Casablanca (on site) · Témara (within 48 h)
1 year warranty — free repair in Morocco if it breaks down during the warranty period
100% secure payment: Bank transfer · Bank card · Cash on delivery (deposit of 300 DH by card or bank transfer) · Certified cheque
Product overview
Give your telephony a boost with the Grandstream HT802 two-port adapter. Enjoy outstanding call quality with its two FXS ports, TLS and SRTP security encryption, and automatic configuration options. Ideal for home and business environments, this adapter supports 3-way conferencing and T.38 fax. Its compatibility with Grandstream's UXM range of IP PBXs makes integration easy and simplifies the management of your VoIP communications.
Request a quoteShare:
Technical specifications
| Display | None |
|---|---|
| Dimensions (W x D x H) | 100 x 100 x 29.5 mm (dimension order not published by the manufacturer) |
| SIP accounts | 2 SIP profiles, one per FXS port |
| Voice codecs | G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.722, G.723.1, G.729A/B, G.726-32, iLBC, OPUS |
| Fax over IP | T.38-compliant fax relay, Group 3 up to 14.4 kbps, auto-switch to G.711 for fax pass-through |
| VoIP protocols | SIP (RFC 3261), over TCP and TLS |
| Call management | Caller ID display or blocking, call waiting, flash, blind or attended transfer, call forwarding, call hold, do not disturb |
| Ringer load | 2 REN, up to 1 km on a 24 AWG line |
| Caller ID | Bellcore type 1 and 2, ETSI, BT, NTT and DTMF-based CID |
| Local conferencing | 3 participants |
| Supported network protocols | TCP/IP/UDP, RTP/RTCP (RFC 1889, 1890), HTTP/HTTPS, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SSH, Telnet, STUN (RFC 3489, 5389), SRTP, SNMP, TR-069, IMS/3GPP, IPoE |
|---|---|
| Network interfaces | 1 auto-sensing 10/100 Mbps Ethernet port (RJ45) |
| Telephony interfaces | 2 FXS ports for analogue phones |
| Weight | 114 g |
| Security | SRTP for media, TLS/SIPS/HTTPS for control, secure automated provisioning with AES encryption, remote management via syslog and SSH |
| Power supply | Power adapter 100-240 V AC 50/60 Hz to 5 V DC / 1 A |
| Device type | ATA adapter |
| FXS ports | 2 FXS ports |
| Management software | GDMS (Grandstream Device Management System) |
| Warranty | 1 year |
Specific references
| EAN | 6947273702047 |
|---|
Downloads
Customer reviews
No reviews yet
Be the first to review this product and help other customers.
Write a review
Frequently asked questions about this product
Can two phones be connected with two different numbers?
Yes. The HT802 has two FXS ports and handles two SIP profiles: each analogue phone is linked to its own account, and therefore to its own line. Both share a single 10/100 network port and benefit from the same features: blind or attended transfer, call forwarding, call hold, do not disturb and 3-way conferencing.
Can I keep using my fax machine when switching to VoIP?
Yes. The HT802 supports T.38 Group 3 fax relay at up to 14.4 kbps, with auto-switching to G.711. Voice, fax and modem are among the supported devices: a fax machine already in service can therefore stay in place after the switch to an IP line.
Does an old rotary-dial phone work with it?
Yes: both dialling modes are accepted, DTMF (tone) and pulse. The adapter also handles caller ID in Bellcore Type 1 and 2, ETSI, BT, NTT and DTMF formats, and disconnection by busy tone, polarity reversal or loop current — enough to cover a mixed set of phones.
How far from the adapter can the analogue phone be placed?
Up to 1 km on a 24 AWG line, for a ringer load of 2 REN. Enough to serve a phone at the other end of a building, in a workshop or a warehouse, reusing the telephone cabling already in place.
How does it connect to the network, and how can I tell it is working?
Via an auto-sensing 10/100 Mbps RJ45 Ethernet port, to connect to a free port on your internet modem, router or switch. Four indicator lights — power, Internet, phone 1, phone 2 — show the status of the device line by line, and a factory reset button lets you start again from scratch.
What happens if the primary SIP server goes down?
It switches automatically to the secondary SIP server as soon as the primary loses its network connection. Telephony on the site then resumes without manual intervention, which greatly limits how long calls are cut off.
Are calls through the HT802 encrypted?
Yes: signalling is protected by TLS/SIPS and voice by SRTP. Administration is done over HTTPS or SSH, and auto-provisioning uses AES encryption, which also protects account credentials. 802.1Q/802.1p, ToS and DiffServ QoS also lets you prioritise voice on the network.
Can several be deployed without configuring them one by one?
Yes. The HT802 supports auto-provisioning via TR-069 and XML files, and provisions itself with Zero Configuration behind an IP PBX from the Grandstream UCM range. The GDMS platform then lets you manage, monitor and update devices in batches, free of charge.
How does it avoid the echo and dropouts typical of voice over IP?
With a dynamic jitter buffer and advanced line echo cancellation, two mechanisms that correct the defects specific to an IP link. On the codec side: G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.722, G.723.1, G.729A/B, G.726-32, iLBC and OPUS.
Is the adapter new and under warranty?
Yes, new and supplied with its invoice, which serves as the warranty document for 1 year. Its universal power supply accepts 100 to 240 V, 50/60 Hz (5 V / 1 A output) and it complies with FCC 15B, EN 55022, EN 55024 and EN 60950. Full details in the datasheet (PDF).
