Grandstream HT818 configuration to FreePbx

I have been trying with no success to connect this 8 port FXS to my freePbx. Is there documentation some where on how to? I have searched the web with little success.


What troubles are you having? If you are new to this, I suggest setting up one extension. Once working, you can make up to 7 more like it.

Create the extension in FreePBX. If you’re not confident that you’ve done it correctly, test it with a softphone or another device which you know how to configure.

GS manual is at

On the Profile 1 page, set Profile Active and enter the IP address of your PBX in Primary SIP Server. See Screenshot on p.62.

On the FXS ports page (see p.63) enter the extension number in both SIP User ID and Authenticate ID, enter the Secret for the extension in Password, Apply and test.

The Status page will show whether it has successfully registered.

If you still have trouble, post details.

No success.

Extension shows as offline and on the HT818 it is unregistered

Do I need to create a SIP trunk that points to the HT818?

Please confirm that with another device or app, you were able to register to the pjsip extension you created, and make/receive calls.

If this is an on-site PBX, confirm that the HT and PBX are on the same subnet. If a cloud PBX, confirm that the device used to test the extension was on the same network as the HT.

Some devices are not compatible with the long passwords autogenerated by FreePBX; possibly the HT is one of them. Create a short password (no more than 12 random letters and digits) and paste it into both Secret for the extension and Password in the HT.

If you still have trouble, report what appears in the Asterisk log when the HT attempts to register. If nothing, run sngrep and report what if anything appears there when the HT attempts to register.

It’s possible to set up the HT as a trunk, but you would lose all extension features (voicemail, DND, etc.) This may make sense if you have e.g. a bank of fax machines.

Am able to place and receive calls on softphone - - GW
All my IP sets are also on the same subnet as the softphone and able to make and take calls

HT818 - GW

site to site VPN

FreePbx - GW

When I watch the logs I seen no activity from the HT

Confirm that there is no NAT involved in the VPN setup.

Based on what I know so far, I would assume that the netmask on both ends should be
Otherwise, a packet sent from to would ARP for the destination and the router would have to do proper proxy ARP magic.

But in any case, I don’t see why you would use different netmasks for and

Likewise, why don’t you have for the netmask on the PBX?

However, in Asterisk SIP Settings, Local Networks should be / 16, so that Asterisk will know not to substitute its public IP address in the Contact header and SDP sent to another site. If you change this, restart Asterisk.

If the network stuff is set up correctly (and there is no NAT), from a shell prompt on the PBX you should be able to ping ; if not, this is a simple networking issue that should not be hard to track down.

If you mean just the Asterisk logs, run sngrep (or tcpdump) to determine whether REGISTER requests are reaching the PBX and what replies, if any, are being sent out. These tools see incoming packets before the FreePBX firewall and outgoing after, and should show if the firewall is blocking anything. Possibly, a previous error caused the HT’s address to be banned; check Intrusion Detection.

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