Question about Grand stream HT801 v2

Asterisk Asterisk Endpoints
1

Hello,

I developed a PBX with asterisk using two linksys SPA3102’s and 2 analog phones, I have deployed both in different areas and testing went well. I am now looking to expand by replacing these outdated ATA’s with new models. From what I have read the best option would be a GS HT801 v2 as its much cheaper than Cisco ATA’s and I only need 1 phone line.

My question is, if I get the GS HT801 v2 will it work with my Asterisk server? I have read online that some ATA’s come locked to a VoIP provider, which I do not have, just a windows machine running my Asterisk server. I apologize if this is an obvious question as I am very new to all this and want to make sure I buy something that will work for what I need!

TLDR: Will a Grandstream HT801 V2 work with my own Asterisk server, or will it require a VoIP provider / using Grandstreams UCM series PBX’s?

Thanks!

2

my experirnce it is open and configurable

3

I am using Grandstrem HT802 purchased from distribution just fine on my Asterisk 22.10 box.

If you buy second-hand on Ebay you never quite know what you are getting.

Mike

4

My first question would be if you need analog service at all.

Just in case you’re not aware, you can port your existing numbers to a SIP provider and calls will be sent to your Asterisk server over the Internet with no hardware required. This is usually way cheaper than service from your local telco or ‘bundled’ service from your cable TV provider.

If you’re not ‘committed’ to the current local numbers, you can rent DIDs almost anywhere in the world and have calls terminate at your Asterisk server. Impress your friends with the number to your ‘Paris office’ or get a DID local to your parents so they only have to make a local (to them) call to you.

5

Or get a few numbers. Numbers are between $1-2/month and minutes are under $0.008. Or should be if you look around. I currently use Twilio,com.

6

First question, what do you mean existing numbers? do you mean the extensions I created on my Asterisk server? From the quick few searches I just did, a DID is just another phone number right?

I appreciate the guidance, I have 150+ analog telephones spread around my state, each one in a different location that can all “talk” (not sure if that’s the right term) on a private IP, I have my windows machine always running an asterisk server in an IDF room which is on a different subnet. Again, not sure if I explained that well at all lol, I am still learning lots!

7

On Wednesday 22 July 2026 at 19:22:52, harrisonM wrote:

First question, what do you mean existing numbers? do you mean the
extensions I created on my Asterisk server? From the quick few searches I
just did, a DID is just another phone number right?

I think sedwards assumed that you had telephones connected to the Public
Switched Telephone Network, which would necessarily mean that they had been
assigned numbers by the local telephone service provider, and it was these
numbers which might be more important to you than the actual hardware devices
which connect to the PSTN.

Maybe this is not the case?

I appreciate the guidance, I have 150+ analog telephones spread around my
state,

Is this a State in America (or a similar country) or are you using the word
“state” in some other sense here?

each one in a different location that can all “talk” (not sure if that’s the
right term) on a private IP,

So, all these phones have ATAs (analogue telephone adapters, which enable them
to communiate using SIP) on some VPN network (thus allowing them to use
private IP addresses and connect to each other)? What ATAs are you currently
using to allow these phones to have IP addresses?

I have my windows machine always running an asterisk server in an IDF room

IDF (disambiguation) - Wikipedia Hm, what?

which is on a different subnet.

So, is there anything interesting about the router connecting these subnets?

Maybe that is the VPN endpoint for all the telephones’ ATAs?

Again, not sure if I explained that well at all lol, I am still learning
lots!

Excellent :slight_smile:

Antony.

–
Just when you think you’re done, a cat floats by with buttered toast strapped
to its back.

  • Steve Krug, “Don’t make me think”
8

It’s my superpower :frowning:

Google says " An IDF (Intermediate Distribution Frame) room is a localized telecommunications closet or server room that acts as a secondary network distribution point."

9

Hi,

what my AI says:

Yes, the HT801 v2 should work perfectly well with Asterisk. It is a standard SIP ATA; you do not need a Grandstream UCM or an external VoIP provider. The UCM integration Grandstream mentions is mainly for their zero-configuration provisioning.

One important distinction when replacing an SPA3102: the SPA3102 has both an FXS port (for an analogue telephone) and an FXO port (for connecting an analogue PSTN line). The HT801 v2 has FXS only.

So if you are currently using the SPA3102 simply to connect an analogue telephone to Asterisk, the HT801 v2 is the correct type of device. If you are using the SPA3102’s LINE/FXO port, the HT801 is not a direct replacement; something like the HT813 would be closer.

From your description of 150+ analogue phones at different sites, it sounds like you only need the FXS function anyway.

Also, the phones themselves do not really have private IP addresses — the ATAs do. If those private networks are routed/VPN-connected to the Asterisk network, there is no inherent problem with them being on different subnets. You just need routing/firewall rules for SIP and RTP between the ATAs and Asterisk.

For 150+ units I would also look into automated provisioning rather than configuring every ATA manually. The HT801 v2 supports XML/TR-069 provisioning, and Grandstream also offers GDMS for centralized management.

As others mentioned, I would buy normal retail/distributor units rather than provider-supplied or unknown used units, since provider-customized devices can potentially be locked or pre-provisioned.

10

hey grate i used some vertical line (just some dashes) to seperate my
own comment and it was completely obmited by the mailinglist
connector…, here again the rest of my mail:

what i can say:

did you reset it to factory and flashed the latest firmware?

sip can be bitchy so try out different codecs, registration, keep alive
and other stuff. It is designed to not work flawlessly between
providers, so maybe an alternative firmware or an h232 connection can
help…

grandstreams in general are fine, i have expirience with them on
asterisk, they work grate and on innovapone, dosnt work correctly with
external calls and with internals only sometimes, otherwise only one
direction (listening not speaking or the outher way around) so i guess
grandstream is most likely not the issue

btw, i would suggest running asterisk on linux, there you are able to
download opencode and let it fix your setup for free (do this on a test
installation, create a snapshot/checkpoint before and if it is not
virtualizized to a full backup) :wink: https://opencode.ai/ maybe also works
on windows but much worse (use veeam backup and restore (free windows
agent) or sth like that there)…

Regards,

TN

Am 13.08.2026 um 23:15 schrieb TN:

Hi,

what my AI says:

Yes, the HT801 v2 should work perfectly well with Asterisk. It is a
standard SIP ATA; you do not need a Grandstream UCM or an external
VoIP provider. The UCM integration Grandstream mentions is mainly for
their zero-configuration provisioning.

One important distinction when replacing an SPA3102: the SPA3102 has
both an FXS port (for an analogue telephone) and an FXO port (for
connecting an analogue PSTN line). The HT801 v2 has FXS only.

So if you are currently using the SPA3102 simply to connect an
analogue telephone to Asterisk, the HT801 v2 is the correct type of
device. If you are using the SPA3102’s LINE/FXO port, the HT801 is not
a direct replacement; something like the HT813 would be closer.

From your description of 150+ analogue phones at different sites, it
sounds like you only need the FXS function anyway.

Also, the phones themselves do not really have private IP addresses —
the ATAs do. If those private networks are routed/VPN-connected to the
Asterisk network, there is no inherent problem with them being on
different subnets. You just need routing/firewall rules for SIP and
RTP between the ATAs and Asterisk.

For 150+ units I would also look into automated provisioning rather
than configuring every ATA manually. The HT801 v2 supports XML/TR-069
provisioning, and Grandstream also offers GDMS for centralized
management.

As others mentioned, I would buy normal retail/distributor units
rather than provider-supplied or unknown used units, since
provider-customized devices can potentially be locked or pre-provisioned.

what i can say:

did you reset it to factory and flashed the latest firmware?

sip can be bitchy so try out different codecs, registration, keep
alive and other stuff. It is designed to not work flawlessly between
providers, so maybe an alternative firmware or an h232 connection can
help…

grandstreams in general are fine, i have expirience with them on
asterisk, they work grate and on innovapone, dosnt work correctly with
external calls and with internals only sometimes, otherwise only one
direction (listening not speaking or the outher way around) so i guess
grandstream is most likely not the issue

btw, i would suggest running asterisk on linux, there you are able to
download opencode and let it fix your setup for free (do this on a
test installation, create a snapshot/checkpoint before and if it is
not virtualizized to a full backup) :wink: https://opencode.ai/ maybe also
works on windows but much worse (use veeam backup and restore (free
windows agent) or sth like that there)…

Regards,

TN

11

and now the part is not omitted any more, this connector clearly is in
the wrong