HT812 not connecting to Internet


My HT812 VOIP Adapter does not connect to Internet. With a verified connection, the blue Internet light flashes on-off. I get a dial tone, but no calls in or out. I did a ‘reset’, but it is still the same. Thanks for any ideas and suggestions.


Post the settings of the device.


Hi Ipneblett, I have cut-copied-pasted the entire settings for Status, Basic Settings, Advanced Settings, Profile 1, Profile 2, and FXS Ports. Let me know what is relevant, and I can post them. Thank you, Bill.


The profile and the FXS settings. You can mask any sensitive IP info. What are you trying to connect the HT to?


I connect the HT812 to the Internet via short LAN cable to my 8-port router. Output is connected to a telco patch panel for analog phones.
Note: The HT812 blue Internet light will flash on-off whether connected to the router, or directly to the CATV modem.

Profile Active: No “Yes”
Primary SIP Server: “BLANK” (e.g.,, or IP address)
Failover SIP Server: “BLANK” (Optional, used when primary server no response)
Prefer Primary SIP Server: “No” Yes ( yes - will register to Primary Server if Failover registration expires)
Outbound Proxy: “BLANK” (e.g.,, or IP address, if any)
Allow DHCP Option 120( override SIP server ): “No” Yes
SIP Transport: “UDP” TCP TLS (default is UDP)
NAT Traversal: “No” Keep-Alive STUN UpnP
DNS Mode: “A Record” SRV NAPTR/SRV
SIP Registration: No “Yes”
Unregister On Reboot: “No” Yes
Outgoing Call without Registration: No “Yes”
Register Expiration: “60” (in minutes. default 1 hour, max 45 days)
Reregister before Expiration: “0” (0-64800. Default 0 second)
SIP Registration Failure Retry Wait Time: “20” (in seconds. Between 1-3600, default is 20)
SIP Registration Failure Retry Wait Time upon 403 Forbidden: “1200” (in seconds. Between 0-3600, default is 1200. 0 means stop retry registration upon 403 response.)
Enable SIP OPTIONS Keep Alive: “No” Yes
SIP OPTIONS Keep Alive Interval: “30” (in seconds. Between 1-64800, default is 30)
SIP OPTIONS Keep Alive Max Lost: “3” (Number of max lost packets for SIP OPTIONS Keep Alive before re-registration. Between 3-10, default is 3)
Layer 3 QoS: “26” SIP DSCP (Diff-Serv value in decimal, 0-63, default 26)
“46” RTP DSCP (Diff-Serv value in decimal, 0-63, default 46)
Local SIP Port: “5060” (default is 5060 for UDP and TCP; 5061 for TLS)
Local RTP Port: “5004” (even number between 1024-65535, default 5004)
Use Random SIP Port: “No” Yes
Use Random RTP Port: “No” Yes
Refer-To Use Target Contact: “No” Yes
Transfer on Conference Hangup: “No” Yes
Disable Bellcore Style 3-Way Conference: “No” Yes (Using star code *23 for 3-way conference)
Remove OBP from Route Header: “No” Yes
Support SIP Instance ID: No “Yes”
Validate Incoming SIP Message: “No” Yes
Check SIP User ID for incoming INVITE: “No” Yes (no direct IP calling if Yes)
Authenticate incoming INVITE: “No” Yes
Authenticate server certificate domain: “No” Yes
Authenticate server certificate chain: “No” Yes
Trusted CA certificates: “BLANK”
Allow Incoming SIP Messages
from SIP Proxy Only: “No” Yes (no direct IP calling if Yes)
Use Privacy Header: “Default” No Yes
Use P-Preferred-Identity Header: “Default” No Yes
SIP REGISTER Contact Header Uses: “LAN Address” WAN Address
SIP T1 Timeout: “0.5 sec”
SIP T2 Interval: “4 sec”
SIP Timer D: “0” (0 - 64 seconds. Default 0)
DTMF Payload Type: “101”
Preferred DTMF method: (in listed order)
Priority 1: “RFC2833”
Priority 2: “SIP INFO”
Priority 3: “In-audio”
Disable DTMF Negotiation: “No” (negotiate with peer) Yes (use above DTMF order without negotiation)
Generate Continuous RFC2833 Events: “No” Yes (RFC2833 events are generated until key is released)
Send Hook Flash Event: “No” Yes (Hook Flash will be sent as a DTMF event if set to Yes)
Flash Digit Control: “No” Yes (Overrides the default settings for call control when both channels are in use.)
Enable Call Features: No “Yes” (if Yes, call features using star codes will be supported locally)
Offhook Auto-Dial Delay: “0” (0-60 seconds, default is 0)
Proxy-Require: “BLANK”
Use NAT IP: “BLANK” (used in SIP/SDP message if specified)
Use SIP User-Agent Header: “BLANK”
Distinctive Ring Tone:
“Ring Tone 1” used if incoming caller ID is “BLANK”
“Ring Tone 1” used if incoming caller ID is “BLANK”
“Ring Tone 1” used if incoming caller ID is “BLANK”
Disable Call-Waiting: “No” Yes
Disable Call-Waiting Caller ID: “No” Yes
Disable Call-Waiting Tone: “No” Yes
Disable Connected Line ID: “No” Yes
Disable Receiver Offhook Tone: “No” Yes (ROH tone will not be played after offhook for 60 seconds)
Disable Reminder Ring for On-Hold Call: “No” Yes
Disable Visual MWI: “No” Yes
Do Not Escape ‘#’ as %23 in SIP URI: “No” Yes
Disable Multiple m line in SDP: “No” Yes
Ring Timeout: “60” (10-300, default is 60 seconds)
Delayed Call Forward Wait Time: “20” (Allowed range 1-120, in seconds.)
No Key Entry Timeout: “4” (1-15, default is 4 seconds)
Early Dial: “No” Yes (use “Yes” only if proxy supports 484 response)
Dial Plan Prefix: “BLANK” (this prefix string is added to each dialed number)
Use # as Dial Key: No “Yes” (if set to Yes, “#” will function as the “(Re-)Dial” key)
Dial Plan: “{ x+ | +x+ | *x+ | xxx+ }”
SUBSCRIBE for MWI: “No”, do not send SUBSCRIBE for Message Waiting Indication
Yes, send periodical SUBSCRIBE for Message Waiting Indication
Send Anonymous: “No” Yes (caller ID will be blocked if set to Yes)
Anonymous Call Rejection: “No” Yes
Special Feature: “STANDARD”
Session Expiration: “180” (90-64800. default 180 seconds)
Min-SE: “90” (90-64800. default 90 seconds)
Caller Request Timer: “No” Yes (Request for timer when making outbound calls)
Callee Request Timer: “No” Yes (When caller supports timer but did not request one)
Force Timer: “No” Yes (Use timer even when remote party does not support)
UAC Specify Refresher: UAC UAS “Omit” (Recommended)
UAS Specify Refresher: “UAC” UAS (When UAC did not specify refresher tag)
Force INVITE: “No” Yes (Always refresh with INVITE instead of UPDATE)
Enable 100rel: “No” Yes
Add Auth Header On Initial REGISTER: “No” Yes
Use First Matching Vocoder in 200OK SDP: “No” Yes
Preferred Vocoder: (in listed order)
choice 1: “PCMU”
choice 2: “PCMA”
choice 3: “G723”
choice 4: “G729”
choice 5: “G726-32”
choice 6: “iLBC”
choice 7: “OPUS”
Voice Frames per TX: “2”
G723 Rate: “6.3kbps encoding rate” 5.3kbps encoding rate
iLBC Frame Size: “20ms” 30ms
Disable OPUS Stereo in SDP: “No” Yes (removes “/2” from offer)
iLBC Payload Type: “97” (between 96 and 127, default is 97)
OPUS Payload Type: “123” (between 96 and 127, default is 123)
VAD: “No” Yes
Symmetric RTP: “No” Yes
Fax Mode: “T.38” Pass-Through
Re-INVITE After Fax Tone Detected: “Enabled” Disabled
Jitter Buffer Type: Fixed “Adaptive”
Jitter Buffer Length: Low “Medium” High
SRTP Mode: “Disabled” Enabled but not forced Enabled and forced
Crypto Life Time: Disabled “Enabled”
SLIC Setting: “USA 1 (Bellcore 600 Ohms)”
Caller ID Scheme: “Bellcore/Telcordia”
DTMF Caller ID: “DEFAULT” Start Tone
Stop Tone
Polarity Reversal: “No” Yes (reverse polarity upon call establishment and termination)
Loop Current Disconnect: “No” Yes (loop current disconnect upon call termination)
Loop Current Disconnect Duration: “200” (100 - 10000 milliseconds. Default 200 milliseconds)
Enable Hook Flash: No “Yes”
Hook Flash Timing: In 40-2000 milliseconds range, minimum: “300” maximum: “1100”
On Hook Timing: “400” (In 40-2000 milliseconds range, default is 400)
Gain: TX “0dB default” RX “-6dB default”
Disable Line Echo Canceller (LEC): “No” Yes
Disable Network Echo Suppressor: “No” Yes
Outgoing Call Duration Limit: “0” (0-180 minutes, default is 0 (No Limit) )
Ring Frequency: “20” (15-60 Hz, default is 20 Hz )
Ring Tones (Syntax: c=on1/off1-on2/off2-on3/off3;)
Ring Tone 1: “c=2000/4000;”
Ring Tone 2: “c=2000/4000;”
Ring Tone 3: “c=2000/4000;”
Ring Tone 4: “c=2000/4000;”
Ring Tone 5: “c=2000/4000;”
Ring Tone 6: “c=2000/4000;”
Ring Tone 7: “c=2000/4000;”
Ring Tone 8: “c=2000/4000;”
Ring Tone 9: “c=2000/4000;”
Ring Tone 10: “c=2000/4000;”

User Settings
Port SIP User ID Authenticate ID Password Name Profile ID Enable Port
1 “BLANK” “BLANK” “xxxx” “BLANK” “PROFILE 1” No “Yes”
2 “BLANK” “BLANK” “xxxx” “BLANK” “PROFILE 1” No “Yes”
Port Offhook Auto-dial


From the above, it does not appear that you have done anything other than connect the device to an Ethernet cable. If this is true, then you need to obtain the manual and then program the device accordingly. You can start by connecting an analog phone to one of the FXS ports and then pressing “***” (three *) and listening to the prompts. There are two that you will need to listen to explicitly - the IP address and the WAN access settings. You can advance thru the prompts by pressing *. When you get to the WAN prompt, press 9 and then when it indicates “enabled”, you can reboot the device and then access it to program. Then, using the manual, you can program it.


Thank you, Ipneblett. I had purchased and installed this early August 2017. Programmed it then, and was working ok until recently. Seemed as it stopped recognizing the Internet connection (flashing instead of solid blue light). I eventually did a Master Reset with the rear panel reset, which would now read the original factory defaults as you noticed. With the unit still not recognizing a valid Internet connection (flashing blue light), I get the following through the IVR: IP / WAN 00.0B.82.A2.B0.F4 / SUBNET / GATEWAY / DNS SERVER It seems that I can do little with this until I can get it to recognize the Internet connection.


Two thoughts:

  1. You might check via the telephone interface that DHCP is turned on.

  2. You might try manually setting a local IP address & subnet via the telephone interface and then use the Web GUI to see what’s what.


Thank you Ipneblett and drostoker…I found my problem. I had incorrectly said initially that my internet service was coming from my modem to an 8 port router, then to my devices. I did not have a router, but instead an 8 port switch. I received my Grandstream GWN7000 router and I am now back up running. Apparently, my provider Charter CATV (Spectrum) was providing me with IP addresses for my devices that need an IP address. They (Charter-Spectrum) apparently cut that service off to me without notification. Meantime I learned a hard lesson of Router-vs.-Switch differences, and had been without my VOIP service for about 2 weeks…(not bad since 90+% are spam anyway). In the end, my HT8XX was never a problem.


Glad you got it resolved.