GRANDSTREAM • ANALOG TO SIP • ONE-PORT ATA
Grandstream HT801 1-Port Analog Telephone Adapter
Grandstream HT801 is a compact one-port analog telephone adapter designed to connect one compatible analog telephone or fax device to a SIP-based VoIP environment. It provides one FXS telephone interface, one 10/100 Mbps Ethernet interface, one SIP profile, T.38 fax support, secure SIP/media options, automated provisioning methods and practical call features for small offices, branches, reception points, remote workers, service-provider deployments and other locations where an existing analog endpoint still has a useful role.
This page is written as a model-specific decision and deployment guide rather than a generic HT-series overview. It explains what the HT801 can and cannot do, how it fits with a SIP PBX or hosted voice service, what to verify before ordering, how to plan fax or analog-phone migration, and how to distinguish the original HT801 from the separately documented HT801 v2. Technical values should always be checked against the exact hardware revision and current Grandstream documentation before a rollout.
The quick decision: when does HT801 make sense?
Good fit
Choose HT801 when the requirement is to retain one standard analog endpoint while moving the call service to SIP. Typical examples include one existing analog desk phone, a dedicated fax machine that has been validated for the intended FoIP path, a small reception or back-office line, a remote-site analog extension, or a service-provider installation where one FXS port is sufficient.
Look at another model
If two independently configured analog endpoints are required, compare the HT802 family. If you need several FXS ports, dual network interfaces, routing functions or a different analog-line architecture, consider the wider HT range or a higher-density gateway. If the task is to connect an incoming analog carrier line rather than an analog telephone, an FXO-capable device is required instead of simply adding more FXS ports.
Check before ordering
Confirm the exact hardware revision, SIP platform, provider policy, analog device type, codec and DTMF requirements, fax expectations, network addressing, VLAN/QoS design, security settings and power arrangement. The label HT801 and HT801 v2 should not be treated as interchangeable procurement identifiers even where their core use case is similar.
Grandstream HT801 at a glance
Grandstream’s current HT801 product material also lists three-way voice conferencing, failover SIP server support, broad caller-ID format support, advanced telephony functions and compatibility with Grandstream UCM-series provisioning workflows. These features are useful only when the connected PBX, provider, account configuration and network are designed to use them correctly.
What the HT801 actually does in a VoIP design
An analog telephone adapter sits between an analog endpoint and an IP voice service. The FXS side supplies the electrical and signaling interface expected by a standard analog telephone or compatible fax machine. The Ethernet side connects the adapter to the local IP network. The HT801 then registers a SIP account with a suitable PBX or service and translates the user’s analog call behavior into SIP/RTP traffic. This makes it possible to preserve selected analog equipment without keeping a separate legacy PBX solely for that one endpoint.
That bridge is the HT801’s main job. It is not, by itself, a complete business phone system. Features such as extension numbering, inbound route design, IVR menus, hunt groups, call queues, recording policy, office-hours routing, user permissions, trunk capacity and emergency-call policy belong to the PBX or service-provider layer. For a Grandstream-based on-premises deployment, review the broader Grandstream IP PBX systems options and confirm the expected registration and provisioning method before ordering endpoints.
FXS connects to an analog endpoint. If your project needs to bring an existing analog PSTN or carrier line into a SIP platform, that is an FXO requirement. Plug shape alone does not establish compatibility, so identify whether each cable is an endpoint connection or a carrier-line connection before selecting hardware.
Customer requirement → HT801 capability → practical result
| Customer requirement | Relevant HT801 capability | Practical benefit | Buying / deployment consideration |
|---|---|---|---|
| Keep one analog phone | Single FXS port + one SIP profile | Reuses a suitable analog handset on a SIP voice platform | Confirm the handset is standard analog, not a proprietary digital PBX phone |
| Retain a fax workflow | T.38 fax relay and G.711 fax pass-through behavior | Allows a fax machine to be tested on an IP voice path | Fax reliability depends on PBX/provider/network support; perform end-to-end testing |
| Protect signaling and media | TLS/SIPS and SRTP support | Supports encrypted SIP/media configurations | Both ends must support and be configured for compatible security settings |
| Deploy many single-line adapters | TR-069, XML and automated provisioning methods | Reduces repetitive manual configuration when a management process is in place | Plan credential security, template ownership, change control and firmware policy |
| Improve resilience of SIP registration | Failover SIP server setting | Can switch to a secondary SIP server when designed correctly | PBX/provider architecture, DNS, routing and account design still determine real availability |
| Use familiar call controls | Transfer, forward, waiting, hold, DND and three-way calling features | Keeps useful analog-phone workflows available | Feature codes and behavior depend on PBX/service configuration and handset capability |
Verified technical specifications for the original HT801
The following table summarizes values published in Grandstream’s HT801 product material and current documentation for the original HT801. Because Grandstream also publishes an HT801 v2 as a separate product, procurement teams should record the hardware revision and use the matching guide, firmware and power information rather than assuming every item with “HT801” in the name is identical.
| Telephone interface | One FXS port |
|---|---|
| Network interface | One 10/100 Mbps auto-sensing Ethernet port (RJ45) |
| LED indicators | Power, Internet, Phone |
| Telephony features | Caller-ID display/block, call waiting, flash, blind or attended transfer, forwarding, hold, do not disturb and three-way conference |
| Voice codecs | G.711 with PLC/VAD-CNG support, G.722, G.723.1, G.729A/B, G.726-32, iLBC and Opus, with jitter-buffer and echo-cancellation functions described by Grandstream |
| Fax over IP | T.38 Group 3 fax relay up to 14.4 kbps and automatic switch to G.711 for fax pass-through behavior |
| Ring load / distance | Grandstream lists up to 5 REN and up to 1 km on 24 AWG for the original HT801 |
| Caller ID formats | Bellcore Type 1 & 2, ETSI, BT, NTT and DTMF-based CID |
| DTMF | In-audio, RFC2833 and/or SIP INFO |
| QoS | Layer 2 802.1Q VLAN / 802.1p and Layer 3 ToS, DiffServ and MPLS support as documented |
| Provisioning and management | HTTP/HTTPS, SSH, TFTP, TR-069, automated provisioning and syslog options |
| Security | SRTP for media; TLS/SIPS/HTTPS for supported control and management workflows |
| Power | Input adapter 100–240 VAC, 50–60 Hz; device output requirement listed as 5 VDC / 1 A |
| Operating environment | 0–40 °C operating; -10–60 °C storage; 10–90% non-condensing humidity |
| Dimensions / weight | 100 × 100 × 29.5 mm; approximately 102 g |
For the latest vendor information, consult the official Grandstream HT801 product page and the HT80x user guide. Match the guide to the physical unit and firmware family in front of you.
HT801 vs HT801 v2: a small name difference that matters
Grandstream currently maintains separate product and documentation pages for the original HT801 and HT801 v2. Both target the same broad problem—connecting one analog endpoint to SIP—and both are described with one FXS port, one SIP profile and a 10/100 Mbps Ethernet interface. That similarity makes it easy for a purchase request, support ticket or configuration note to omit the revision.
Treat the revision as part of the product identity. Before ordering or replacing a unit, ask for a label photo or exact model reference, confirm the vendor page that applies to it, and use the matching firmware and administration documentation. This is particularly important when a project includes an existing installed base because migration work often mixes units purchased in different years.
| Question | Original HT801 | HT801 v2 |
|---|---|---|
| Core use case | One analog endpoint to SIP | One analog endpoint to SIP |
| FXS ports | 1 | 1 |
| Network interface | 1 × 10/100 Mbps | 1 × 10/100 Mbps |
| Procurement rule | Use original HT801 documentation | Use HT801 v2 documentation and current v2 firmware guidance |
If your requirement is simply “one analog phone to SIP,” either revision may appear functionally close at a high level, but a procurement or support decision should still be based on the exact unit being supplied. Fourteck can help document the model/revision, planned PBX, endpoint and network requirements before quotation or rollout.
Where one FXS port is genuinely useful
1. One analog phone that still serves its purpose
Not every analog phone needs to be replaced just because the PBX or trunking platform moves to SIP. A simple handset at a warehouse desk, back office, security desk, workshop, small branch or service counter may only need dependable inbound and outbound calling. If the user does not need a large display, programmable BLF keys, integrated directory search or a modern headset workflow, retaining the analog handset through an HT801 can be operationally sensible.
2. A single fax machine during a controlled migration
Organizations may still receive forms, confirmations or legacy documents by fax. The HT801 supports T.38 and G.711 fax pass-through behavior, but successful faxing is not guaranteed by the adapter alone. The SIP provider, PBX, codec negotiation, packet loss, jitter, NAT handling and fax machine configuration all affect results. Test representative inbound and outbound pages, longer documents and repeat transmissions before retiring the old service.
3. A branch that needs one analog extension
A branch may use modern SIP phones for most staff but still have one analog requirement. The HT801 allows that one endpoint to register to the same voice environment, subject to network reachability and service policy. This keeps the analog exception small and avoids deploying a multiport gateway where only one port is needed.
4. Service-provider or multi-site standardization
Automated provisioning and remote-management options are valuable when the organization has many sites, each requiring one analog connection. Standardization should include device naming, credential generation, configuration templates, firmware policy, syslog/monitoring practices, replacement procedure and documentation ownership. The device can support these workflows, but the operational system around it determines whether a large rollout remains manageable.
Where HT801 is not the right answer
Two separate analog extensions: one FXS port is not enough. Compare HT802 or another multiport model.
Several analog phones that need independent accounts: use suitable multi-FXS equipment or a higher-density analog gateway based on scale.
An incoming analog PSTN line: this requires an FXO interface. HT801’s FXS port is for an analog endpoint.
A proprietary digital PBX handset: an RJ-style connector does not make a digital proprietary phone analog. Verify the handset/interface type.
Advanced desktop user experience: users who need BLF, large directory views, video, native application integration, multiple line appearances or rich on-phone controls may be better served by a SIP desk phone.
Specialist life-safety or regulated analog equipment: alarms, lift phones, medical devices, payment terminals and similar systems need vendor/application approval and should not be assumed compatible merely because they use an analog connector.
PBX and SIP-provider compatibility: what to confirm
The HT801 uses standard SIP concepts, but a successful project depends on the service on the other side. Before deployment, confirm whether the hosted PBX, on-premises PBX or SIP provider allows third-party ATAs and whether it supplies the required registration credentials. Some services use automatic device onboarding while others require manual SIP authentication. Managed services may restrict device models or provisioning access.
Record the SIP server, optional outbound proxy, authentication ID, user/extension ID, password handling method, transport, registration lifetime, codec preference, DTMF method and any NAT traversal requirements. If a failover SIP server will be used, test the failure and recovery process instead of assuming the secondary value alone provides service continuity.
For Grandstream UCM deployments, Grandstream lists the HT801 as compatible with UCM-series Zero Configuration provisioning. Treat that as a provisioning convenience, not a substitute for dial-plan, extension, trunk, security and network design. If a UCM is part of the project, document the UCM model, firmware, extension policy and whether the ATA will be on the same trusted network or across a routed/WAN connection.
Network planning for voice quality and predictable support
The Ethernet port is only 10/100 Mbps, which is more than sufficient for the bandwidth of a single voice session, but link speed is not the same thing as voice quality. Latency, jitter, packet loss, congestion, NAT behavior, DNS reliability and routing stability matter more. Connect the HT801 to a managed business network where the voice path can be understood and supported.
If the site uses voice VLANs, verify whether the intended switch port and ATA configuration will place the device on the correct subnet. The HT801 documentation lists 802.1Q VLAN and 802.1p QoS capabilities along with Layer 3 ToS/DiffServ/MPLS options. The wider network must honor the design for those markings to have practical value. Avoid enabling QoS values without a matching switch/router policy and avoid introducing a new VLAN without DHCP, DNS, gateway and firewall rules being ready.
For remote sites, confirm the route to the SIP platform, firewall session handling, NAT timers and whether the provider requires specific ports or SBC behavior. Do not expose the management interface directly to the public Internet unless the architecture explicitly calls for it and appropriate controls are in place. Where secure signaling/media are required, validate TLS/SRTP support end to end.
Fax-over-IP planning with HT801
Fax is one of the most common reasons businesses retain an analog endpoint, yet it is also one of the areas where careful testing matters most. The HT801 supports T.38 Group 3 fax relay and can fall back to G.711 fax pass-through behavior. T.38 is intended to make fax transport more tolerant of IP-network conditions by handling fax as fax data rather than simply carrying modem tones as audio.
The complete path must support the selected method. A local ATA with T.38 enabled does not guarantee that the SIP provider, session border controller, carrier interconnect or receiving side will negotiate it. If the path falls back to G.711, packet loss and jitter can affect reliability. Before migration, note the fax machine model, expected page volumes, destinations, whether international faxing is used and whether inbound fax numbers can be tested without business disruption.
Send and receive short and multi-page documents, repeat tests to several destinations, and record whether T.38 negotiated or pass-through was used.
Observe packet loss, jitter and call stability during fax sessions, especially across WAN, VPN or Internet links.
If fax is business-critical, define an alternative document-delivery method instead of relying on an untested IP-fax path.
Security and provisioning considerations
Grandstream documents TLS/SIPS, SRTP and HTTPS support for the HT801, along with provisioning and management methods such as HTTPS, SSH, TFTP and TR-069. Those features provide building blocks for a secure deployment, but secure outcomes depend on configuration. Use unique SIP credentials, change administrative credentials, restrict who can reach the management interface, prefer encrypted management where supported by the provisioning platform, and protect configuration files from unauthorized access.
In a larger rollout, treat provisioning templates as sensitive infrastructure. They may contain server addresses, account details and policies that would be valuable to an attacker. Maintain version control or an equivalent change process, document which template applies to which hardware revision, and test firmware/configuration changes on a small group before broad deployment.
Do not confuse media encryption with complete service security. TLS protects SIP signaling only when the PBX/provider side is configured compatibly, and SRTP protects media only when both ends negotiate it. Firewall policy, administrator authentication, endpoint firmware, DNS integrity, remote-access design and monitoring also form part of the overall voice-security posture.
A practical HT801 installation workflow
Confirm the endpoint. Identify the analog phone or fax machine and verify that it is an appropriate analog device for an FXS port.
Confirm the exact HT801 revision. Record the label/model details and match them to the correct Grandstream documentation.
Prepare the SIP account. Obtain approved server, extension/account, authentication, transport, codec and DTMF information from the PBX or provider administrator.
Prepare the network port. Confirm switch port, VLAN, DHCP/static addressing, DNS, gateway, QoS and firewall expectations.
Connect the analog endpoint. Use an appropriate RJ11 cable from the phone/fax to the HT801 FXS port.
Connect Ethernet. Link the HT801 network port to the approved router or switch path.
Apply the correct power source. Use the power arrangement specified for the exact hardware. The original HT801 documentation lists a 5 VDC / 1 A output supply.
Discover the IP address and manage the unit. Grandstream documentation supports using the analog handset IVR to identify the current IP address, then accessing the embedded web interface from an appropriate management network.
Configure or provision SIP. Apply the approved settings manually or through the organization’s provisioning system.
Validate call direction and features. Test inbound and outbound calling, caller ID, DTMF, hold/transfer behavior, voicemail indication and any required feature codes.
Test failure cases. If failover SIP server, alternate Internet routing or other resilience controls are in scope, test them deliberately.
Document and hand over. Record device location, revision, MAC/asset reference where appropriate, extension, management ownership, firmware policy and support contact path.
Commissioning tests that matter more than “it registered”
A green registration state is only the start. A useful acceptance test should confirm two-way audio, caller ID, inbound and outbound dialing, DTMF through IVRs, transfer behavior, call waiting if required, voicemail access and recovery after a network interruption. If the endpoint is a fax machine, add the fax tests described earlier. If the device is remote, test from its actual WAN path rather than only on a bench network.
| Test | Why it matters | Typical evidence |
|---|---|---|
| Inbound call | Confirms routing, ringing and answer path | Correct extension rings and audio is bidirectional |
| Outbound call | Confirms dialing and trunk/service routing | Expected CLI and destination reachability |
| DTMF to IVR | Catches DTMF-method mismatches | Menu digits are recognized consistently |
| Transfer/hold | Validates user workflow and PBX feature interaction | Attended/blind behavior matches the approved process |
| Power/network restart | Checks automatic recovery | Device reacquires network and SIP registration without manual intervention |
| Failover server | Validates resilience claims in the actual design | Documented switch to secondary service and recovery behavior |
HT801, HT802 or a larger Grandstream ATA?
Start with the number and type of analog connections rather than the model name. One analog endpoint points toward HT801. Two independently used analog endpoints point toward HT802. More endpoints may justify HT814/HT818-class equipment or a higher-density analog gateway. If analog carrier lines are involved, include FXO-capable products in the comparison.
| Model family | Analog endpoint capacity | Best starting point | Do not assume |
|---|---|---|---|
| HT801 / HT801 v2 | 1 FXS | One analog phone or fax endpoint | Revision/firmware are interchangeable |
| HT802 / HT802 v2 | 2 FXS | Two analog endpoints | One SIP account can automatically serve every required workflow |
| HT812 / HT814 / HT818 families | Multiple FXS ports depending on model | Several analog endpoints and, on applicable models, additional network/routing capability | A multiport ATA is always better than multiple small adapters; compare wiring, resilience and support model |
| FXO-capable HT models/gateways | Varies | Integrating analog carrier lines | FXS and FXO are substitutes for each other |
For the wider selection logic, use Fourteck’s Grandstream HT Analog Telephone Adapter Series guide. It focuses on choosing among one-, two- and multiport endpoint adapters and FXO-equipped options without duplicating the model-specific detail on this page.
Dubai and UAE deployment considerations
In Dubai and the wider UAE, HT801 requirements usually appear as part of a PBX refresh, SIP migration, branch standardization, fax-line replacement or retention of one analog endpoint inside a mostly IP-based environment. The key decision is not the city name but the surrounding infrastructure: which PBX or hosted service owns the extension, how the office network is segmented, whether the endpoint is business-critical and who will support it after deployment.
Dubai offices and business centres
A compact office may keep one analog device while the rest of the users move to SIP phones or soft clients. In multi-tenant buildings and business centres, coordinate the local LAN port, DHCP/VLAN assignment and voice-service ownership before installation. If Internet access is shared or managed by another provider, confirm that the SIP path and any required remote provisioning are permitted.
Abu Dhabi enterprise and branch environments
For a branch connected to a central PBX, document WAN routing, DNS, QoS and failover behavior. A single ATA can be simple at the desk but still depend on a larger enterprise network. If the site uses an SD-WAN or VPN overlay, include voice traffic in the network acceptance test rather than treating the HT801 as an isolated endpoint.
Sharjah and Ajman small-business deployments
For smaller offices, the most common mistake is buying an ATA before checking the voice provider. Confirm the SIP account can be configured on a third-party ATA and whether the provider supports the required codecs, DTMF and fax method. If only one analog endpoint remains, HT801 can keep the design economical and easy to support; if two or more endpoints are already known, choose the correct port count from the start.
Fourteck can assist UAE buyers with requirement clarification, model comparison, quotation preparation, compatibility discussion and deployment planning. Price, stock, delivery timing, warranty terms and exact hardware revision should be confirmed at quotation/order stage rather than assumed from a product description.
GCC and Africa project planning
The HT801 is also relevant for distributed organizations that standardize voice equipment across the GCC or African branches. A one-port ATA is easy to ship and deploy, but cross-country consistency requires more than matching hardware. Document local SIP/provider requirements, power accessories, Internet quality, remote-management access, replacement logistics and the team responsible for configuration backups.
For Kuwait or other GCC projects, confirm the service provider’s endpoint policy and whether the same SIP configuration pattern used in the UAE applies. For Uganda, Kenya, Tanzania, Ghana or Nigeria deployments, pay particular attention to branch connectivity quality, backup Internet options and remote support. A small device at a distant site benefits from clear labeling, a repeatable reset/reprovision procedure and a support checklist that local staff can follow without exposing credentials.
Where regional procurement is part of a larger Fourteck project, related country resources can be used for local IT and communication requirements: Kuwait, Uganda, Kenya, Tanzania, Ghana and Nigeria. Confirm local product availability and support scope for each project rather than assuming identical commercial conditions across countries.
Common HT801 deployment mistakes
Mistaking FXS for FXO
An FXS port serves an analog endpoint. It is not an analog carrier-line input. Identify the line type before ordering.
Ignoring the hardware revision
Original HT801 and HT801 v2 have separate documentation. Record the exact unit and use its matching firmware and guide.
Assuming fax will just work
T.38 support is valuable, but the complete provider path must be compatible. Test the actual business workflow.
Treating registration as acceptance
A registered account can still have one-way audio, DTMF, transfer, caller-ID or routing problems. Run a full call-flow test.
Using weak management controls
Change administrative credentials, restrict management access and protect provisioning files and SIP credentials.
Buying the wrong port count
If a second analog endpoint is already planned, one-port hardware creates an avoidable second purchase and extra management point.
Troubleshooting guide for first-line support
The analog phone has no dial tone
Check power, the RJ11 connection, the phone itself and whether the FXS port is enabled. Confirm the endpoint is a standard analog phone. If the unit is registered but there is no usable analog behavior, compare the configuration with the approved template and inspect status/LED indications.
The HT801 does not register
Verify IP addressing, gateway and DNS first, then SIP server reachability, credentials, authentication ID, transport and provider policy. Check whether a firewall or NAT device is blocking or rewriting the SIP path. Registration failures often come from account or network policy rather than a hardware fault.
Calls connect but audio is one-way
One-way audio usually points to RTP/NAT/firewall routing. Confirm the correct media addresses are advertised, the required RTP path is permitted, and any SBC/NAT traversal method is configured consistently. Do not solve it by exposing the device management interface to the Internet.
IVR digits are not recognized
Compare DTMF settings between the HT801, PBX and provider. Grandstream documents in-audio, RFC2833 and SIP INFO methods. A mismatch can allow calls to connect while automated menus fail.
Fax is intermittent
Confirm whether T.38 is negotiated end to end, review codec and fax settings, and measure network loss/jitter. Test several destinations and larger documents. If pass-through is used, the network path must carry modem audio with very little disruption.
The device does not recover after an outage
Check DHCP/DNS availability, SIP registration timers, upstream switch/router recovery and power stability. If a failover SIP server is configured, verify that the secondary target is reachable and authorized for the same account.
Procurement checklist for Grandstream HT801
Exact requested model and hardware revision: HT801 or HT801 v2.
Quantity and whether each site needs one or more analog endpoints.
Analog endpoint type: standard phone, fax machine or another approved device.
PBX or hosted voice platform name and model/service.
SIP provider policy for third-party ATAs.
Need for T.38 fax and business-criticality of fax workflow.
Network topology, VLAN/QoS policy, DHCP/static addressing and firewall constraints.
Security requirement for TLS/SRTP and protected management access.
Provisioning method: manual, UCM Zero Config, TR-069 or other controlled process.
Installation country/site and required power/accessory confirmation.
Support scope: supply only, configuration assistance, deployment planning or post-install troubleshooting.
Quotation confirmation for price, availability, delivery, warranty terms and exact supplied revision.
Why involve Fourteck in an HT801 project?
The value of an ATA project is usually not in the physical connection alone; it is in making the analog endpoint behave correctly inside the new voice environment. Fourteck can help buyers define whether HT801 is the right port count, compare it with HT802 or larger Grandstream options, identify PBX/SIP compatibility questions, plan VLAN and network requirements, document fax testing and prepare a clearer bill of materials before ordering.
For an existing installation, share the current PBX model, provider, device label/revision, analog endpoint type and the problem you are trying to solve. For a new deployment, share the number of analog endpoints, site locations, expected calling features, fax requirement and network design. This information helps separate a simple one-port use case from requirements that need a different ATA, FXO gateway, SIP phone or PBX change.
Fourteck does not need to force every requirement into one product. If HT801 is too small, the comparison should move to a suitable multiport ATA. If the endpoint needs modern SIP-phone features, a desk phone may be the better answer. If the connection is an analog carrier line, an FXO-capable solution should be considered. The objective is a supportable voice design rather than a model-name match.
Grandstream HT801 FAQ
What is the Grandstream HT801?
The Grandstream HT801 is a one-port analog telephone adapter that connects one compatible analog phone or fax endpoint to a SIP-based VoIP system through one FXS port and one 10/100 Mbps Ethernet port.
How many analog phones can the HT801 connect?
It provides one FXS port and is intended as a one-port ATA. If two independent analog endpoints are required, compare the HT802 family or another suitable multiport model.
Does HT801 support fax?
Yes. Grandstream documents T.38 Group 3 fax relay and G.711 fax pass-through behavior. The provider, PBX and full network path must also support the intended fax method, so end-to-end testing is recommended.
Can HT801 work with a Grandstream UCM PBX?
Grandstream lists HT801 for use with UCM-series IP PBXs and supports Zero Configuration provisioning workflows. Confirm the exact UCM model, firmware, extension policy and network design for the project.
Does HT801 support encrypted VoIP?
The HT801 supports TLS/SIPS control options and SRTP media security. Encryption must be configured compatibly on the PBX or SIP-provider side as well.
What Ethernet interface does the original HT801 use?
Grandstream specifies one 10/100 Mbps auto-sensing RJ45 Ethernet interface for the original HT801.
What is the difference between HT801 and HT801 v2?
They are separately documented Grandstream hardware revisions with a similar one-port ATA role. Treat the revision as part of the product identity and use the matching product page, firmware and administration guide for the actual unit.
Can HT801 connect an analog telephone line from a telecom provider?
HT801 provides an FXS port for an analog endpoint. Bringing an analog carrier/PSTN line into a VoIP system is an FXO requirement and needs appropriate FXO-capable equipment.
Does HT801 support VLAN and QoS?
Grandstream documentation lists 802.1Q VLAN, 802.1p and Layer 3 QoS-related capabilities. The switch, router and wider network must be configured consistently for those settings to provide a useful result.
Can HT801 be provisioned remotely?
Grandstream documents automated provisioning methods including TR-069 and XML configuration workflows, as well as common management protocols. Protect provisioning credentials and restrict management access.
What power does the original HT801 require?
Grandstream’s original HT801 datasheet lists a universal 100–240 VAC input power adapter with 5 VDC / 1 A output. Confirm the exact hardware revision and included power accessory before deployment.
Is HT801 suitable for a Dubai office?
Yes when the office needs one compatible analog endpoint connected to a supported SIP/PBX platform. Confirm the UAE voice-provider policy, network configuration, required revision, availability and support scope before ordering.
Should I choose HT801 or HT802?
Choose by port requirement. HT801 is the one-port option; HT802 is designed for two FXS endpoints. Also verify the exact original/v2 revision and network features required.
Can Fourteck configure the HT801 for my PBX?
Fourteck can discuss configuration and deployment requirements based on the PBX, SIP provider, network and support scope. Share the platform details and required call/fax behavior so compatibility and service scope can be confirmed.
How do I request an HT801 quotation?
Send Fourteck the required quantity, destination, exact model/revision if known, PBX or provider, analog endpoint type and any configuration or installation requirement. Price, availability, delivery and warranty terms can then be confirmed for the quotation.
Plan the analog-to-SIP connection before you order
If your requirement is one analog phone or fax endpoint on a SIP service, the Grandstream HT801 is a focused, compact option. The best next step is to confirm the exact revision, PBX/provider, endpoint type, network and fax requirements. That prevents common errors such as ordering FXS for an FXO need, buying too few ports, or treating HT801 and HT801 v2 as the same support item.
Send Fourteck your current voice platform, destination country, number of analog endpoints and required features. We can help compare the right HT model, identify compatibility questions and prepare the deployment or quotation discussion without assuming stock, price or commercial terms that still need confirmation.
Manufacturer references
Technical details on this page were structured around Grandstream’s official HT801 product page, HT801 datasheet and current HT80x documentation. Because firmware and product revisions can change, use the official vendor documentation for the exact hardware in your hand during configuration and troubleshooting.





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