Grandstream HT841 - Setup Guide
FXO Gateway
Table of Contents
HT841 FXO (Foreign Exchange Office) Gateway Setup Guide Step 1: Create the Domain Step 2: Create the SIP Trunk Step 3: Add the DID to Inventory Step 4: FXO Profile 1 — GeneralCloud PBX Settings Step 5: FXO Profile - FXO Termination Settings Step 6: FXO Ports Configuration Step 7: Confirm Registration Step 8: Inbound Call Test Step 9: Outbound Call TestHT841 FXO (Foreign Exchange Office) Gateway Setup Guide
Platform: Hosted PBX | Device: Grandstream HT841 (4-port FXO ATA) | Updated: May 2026
This guide covers the end-to-end setup of a Grandstream HT841 to bridge a PSTN analog line into the Hosted PBX platform. The workflow follows three phases:
Phase 1: Platform Provisioning (domain, SIP trunk, DID)
Phase 2: HT841 Device Configuration
Phase 3: Validation
Phase 1Platform Provisioning
Step 1: Create the Domain
Log in to the reseller portal and navigate to Domains → Add Domain.
- Set the domain name (e.g.,
acmetrunk.sip) - Assign the appropriate reseller/territory
- Save and confirm the domain is active
Step 2: Create the SIP Trunk
Under the new domain, navigate to Trunks → Add Trunk.
| Field | Value |
|---|---|
| Trunk Type | Registration-based (for HT841 outbound registration) |
| SIP Username / Auth ID | acmetrunk |
| SIP Password | Generate a strong password |
| From Domain | acmetrunk.sip |
| SIP Proxy Address | <Sip Server URL Here> example sip.siplogin.ca |
| Port | 6060 |
Step 3: Add the DID to Inventory
Navigate to DIDs / Phone Numbers → Add DID.
- Enter the customer's full E.164 DID (e.g.,
15879971212) - Assign it to the domain and trunk created above
- Set the inbound routing destination (hunt group, extension, auto-attendant, etc.)
- Save and confirm that the DID appears as active in the inventory
Phase 2HT841 Device Configuration
Log in to the HT841 web UI (default: http://<device-ip> The admin password is printed on the device label.
Step 4: FXO Profile 1 — GeneralCloud PBX Settings
| Setting | Value |
|---|---|
| Profile Active | Yes |
| Primary SIP Server | acmetrunk.sip |
| Failover SIP Server | Leave blank |
| Prefer Primary SIP Server | No |
| Outbound Proxy | <Sip Server URL Here> example sip.siplogin.ca |
| Backup Outbound Proxy | Leave blank |
| Prefer Primary Outbound Proxy | No |
| From Domain | acmetrunk.sip |
| Layer 3 QoS SIP DSCP | 26 |
| Layer 3 QoS RTP DSCP | 46 |
| DNS Mode | A Record |
Click Save before moving to the next tab.
Step 5: FXO Profile - FXO Termination Settings
| Setting | Value |
|---|---|
| Enable PSTN Disconnect Tone Detection | No |
| Enable Polarity Reversal | No |
| AC Termination Model | Country-based |
| Country-based | USA |
| Number of Rings | 2 (see important note below) |
| PSTN Ring Thru FXS | Yes |
| PSTN Ring Thru Delay (sec) | 4 |
| PSTN Ring Timeout (sec) | 6 |
| PSTN Idle Wait Timeout between Outgoing Calls | 4 |
⚠ IMPORTANT Number of Rings: Set this to 2. Setting it to 1 will cause Caller ID to not be fully transmitted to the SIP server before the call is answered, resulting in missing or incorrect Caller ID on inbound calls.
Click Save.
Port Settings → FXO Profile 1 → Channel Dialing
This enables the ability to allow outbound calling through the analog port. Custom setup required. Please contact support.
| Setting | Value |
|---|---|
| Stage Method | 1 |
| Min Delay Before Dial PSTN Number | 800 |
Step 6: FXO Ports Configuration
Port 1 (active line):
| Field | Value |
|---|---|
| SIP User ID | acmetrunk |
| Authenticate ID | acmetrunk |
| Authenticate Password | Password from Step 2 |
| Profile ID | FXO Profile 1 |
| Hunting Group | Active |
| Request URI Routing ID |
5879971212 (10-digit DID, no leading 1) |
| Enable Port | No (FXO uses registration, not direct enable) |
| Unconditional Call Forward to PSTN | Yes |
Ports 2–4: Leave Hunting Group as Disabled unless additional analog lines are connected to those ports.
NOTE:
If additional ports are required, set Hunting Group to Active on the registered port, and set the remaining required ports to the port number that is registered.
NOTE:
If additional ports are required, set Hunting Group to Active on the registered port, and set the remaining required ports to the port number that is registered.
Unconditional Call Forward to VoIP - Port 1:
| Field | Value |
|---|---|
| CID |
15879971212 (full E.164 with leading 1) |
| SIP Server | <Sip Server URL Here> example sip.siplogin.ca |
| SIP Destination Port | 6060 |
This section controls where inbound PSTN (Public Switched Telephone Network) calls are forwarded when the HT841 receives a ring on an FXO Port. It sends the call to the Hosted PBX platform using the CID value configured above.
Click Save and Apply.
Phase 3Validation
Step 7: Confirm Registration
In the HT841 web UI, navigate to Status → Port Status.
- FXO Port 1 should show Registered against
acmetrunk.sip - If not registered after 60 seconds, double-check the SIP username, password, and proxy address
Step 8: Inbound Call Test
- Call the customer's DID from an external phone
- Confirm the call rings the HT841's FXO Port 1
- Confirm the call is forwarded to the Hosted PBX platform and routes correctly
- Verify Caller ID (CID) is passed correctly if CID shows blank or wrong, recheck the Number of Rings (must be 2 or higher)
Step 9: Outbound Call Test
- From the Cloud PBX extension, place a call out via the PSTN trunk
- Confirm dial tone is heard and the call completes
- Confirm the outbound Caller ID presents the customer's DID correctly
ReferenceQuick Reference — Key Values
| Item | Value |
|---|---|
| SIP Proxy / Outbound Proxy | <Sip Server URL Here> example sip.siplogin.ca |
| SIP Destination Port (VoIP forward) | 6060 |
| SIP Domain / From Domain | acmetrunk.sip |
| Number of Rings (FXO Termination) | 2 |
| CID format — Call Forward to VoIP | E.164 with leading 1 (e.g., 15879971212) |
| Request URI Routing ID | 10-digit DID, no leading 1 (e.g., 5879971212) |
Support Common Issues
| Symptom | Likely Cause | Fix |
|---|---|---|
| No Caller ID on inbound calls | Number of Rings set to 1 | Change to 2 in the FXO Termination settings |
| FXO port not registering | Wrong SIP credentials or proxy address | Re-verify username, password, and outbound proxy |
| Inbound calls are not routing to the Cloud PBX | Call Forward to Cloud PBX is not configured | Check the CID, SIP Server, and Port in FXO Ports |
| Outbound calls failing | Trunk is not active on the platform | Confirm trunk status in the Hosted PBX |