Configuration for Grandstream HT812

[SuperCoolYolo]

Hi there,

Adding to the thread as I think it's probably the most appropriate.

I've just purchased a new Grandstream HT812 ATA and had went through all the setup instructions in these forums. I finally managed to get it working so thought I'd make a post to share the instructions in case anyone wondered like I did. It's not overly complicated for someone who has some basic knowledge of how to setup a home router and network -- many of the settings are just set as per instructions found around this forum, don't even need to know what the settings do. I have the steps below documented according to the latest setting labels to help those who have this device and firmware (the older instructions are still applicable but I'm hoping this makes it clearer).

I have it working in my setup where the ATA is actually behind 3 routers like this:
Internet cable -> Cable Modem (router) -> Mesh router -> Personal router -> HT812 ATA

Here are the steps I took:

Download the latest firmware (version 1.0.29.8 as of this writing) from the Grandstream web site for HT812. Unzip the file to a location on the computer.

Login using web browser to the HT812 device on the local network.

On the Advanced Settings tab:
Scroll to the bottom, for the label "Upload Firmware" click on "Upload from local directory" then on the next screen, click "Browse..." to locate the unzipped .bin file (e.g. ht81xfw.bin) from your computer. Click "Upload Firmware" to upgrade to the latest firmware. Let the device do its thing and settle down, which I think took about 10 minutes for me.

Login again to the HT812 device.

On the Basic Settings tab:
Set the time zone.

On the Advanced Settings tab:
Update the admin password.
Disable the user level & viewer level access (my preference).

To get things going, it's best to leave the other Basic and Advanced Settings as is then tweak it later to meet your needs.

On the Profile 1 tab, set these settings.

Primary SIP Server:
voip.freephoneline.ca
voip2.freephoneline.ca (alternative)

NAT Traversal: Keep-Alive
Outgoing Call without Registration: No
SIP Registration Failure Retry Wait Time: 120 (seconds)
Enable SIP Options Keep Alive: OPTIONS (there isn't an option for 'yes' as per the older forum instructions)
SIP OPTIONS/NOTIFY Keep Alive Interval: 20 (seconds)
Use Random SIP Port: Yes
Use Random RTP Port: Yes
Transfer on Conference Hangup: Yes
Allow Incoming SIP Messages from SIP Proxy Only: Yes
SIP REGISTER Contact Header Uses: WAN Address
Preferred DTMF method:
Priority 1: RFC2833
Priority 2: In-audio
Priority 3: SIP INFO
Enable Call Features: No
No Key Entry Timeout: 4 (seconds)
Preferred Vocoder:
Choice 1: PCMU
Choice 2: G729
Choice 3: PCMU
Choice 4: PCMU
Choice 5: PCMU
Choice 6: PCMU
Choice 7: OPUS (left it as is)
Choice 8: G722 (left it as is)

Click the Apply button. (you may have to reboot the device, I didn't capture that in my notes)

On the FXS Ports tab:
Set the SIP User ID and Authenticate ID fields with your ID assigned by FreePhoneLine
Set the Password assigned by FreePhoneLine
Set the Profile ID to Profile 1 to match what you had set above
Set Enable Port to Yes

Click the Apply button. (you may have to reboot the device, I didn't capture that in my notes)

Go to the Status tab, check under Port Status that your device shows FXS 1 as Registered. Notice FXS 2 shows as Not Registered (unless you've got that configured separately).

That's all, it should just work after inputting the above settings.

Good luck ya'll!

[Liptonbrisk]

Hi there,

Adding to the thread as I think it's probably the most appropriate.

Hello,

You posted in a topic that's almost a year old. I've moved the post to device configuration, since post is about device configuration.

Thank you for your input.

Primary SIP Server:
voip.freephoneline.ca
voip2.freephoneline.ca (alternative)

voip4.freephoneline.ca:6060 may also be used by everyone, regardless of the ISP used, and is intended for those who are having difficulties trying to circumvent SIP ALG.

Enable Call Features: No

If that setting is disabled, you will not be able to use the ATA's star codes as listed on pages 29 to 30, even if they do work in conjunction with FPL: https://www.grandstream.com/hubfs/Produ ... _guide.pdf

Transfer on Conference Hangup: Yes

This option is intended for 3-way calling. When hanging up, the call is supposed to be transferred to the second person in order for the call to continue with the third person.
I don't have this Grandstream ATA to test with FPL, but if the call transfer is only supported by the ATA bridging the calls together, your FPL account may be tied up by allowing the call to continue.
I would test to confirm whether the feature actually works with FPL (that is, whether the call actually continues between the two other parties after you hang up) and also to see whether you can receive further calls after hanging up. If the feature doesn't work, then this setting is inconsequential. If it does work, you may not want it to depending on how the ATA handles the call.
FPL accounts are limited to 2 channels (or calls) per account.

Choice 7: OPUS (left it as is)
Choice 8: G722 (left it as is)

These audio codecs are not supported by FPL at all.

Notice FXS 2 shows as Not Registered (unless you've got that configured separately).

Yes, unless the user has two VoIP unlock keys or another service besides FPL, FXS2 should remain unregistered.
FPL allows only one registration per account at any time. I have seen some users trying to register the same VoIP unlock key more than once simultaneously, which leads to issues with incoming calls.