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I have attached the HT802 config for future users

TankCla wrote:I have attached the HT802 config for future users

1. Please indicate where to enter SIP Username and SIP Password. Thanks.

2. Transfer on Conference Hang-up

Has anyone tested to see whether setting "Transfer on Conference Hang-up" to "Yes", actually works properly with Freephoneline?
I don't have a Grandstream ATA to test.

That is, when set to Yes, during a 3-way call, when the FPL Grandstream user hangs up, the call is supposed to continue between the other two parties.

The default setting is no, which means the call just gets completely dropped when the FPL user hangs up.

If that setting doesn't work with FPL (that is, if the call doesn't continue between the the two other parties when FPL caller hangs up), it should just remain at the default, "No",

http://www.grandstream.com/hubfs/Produc ... _guide.pdf

Transfer on Conference Hang-up

"If set to "Yes", when the phone hangs up as the conference initiator, the
conference call will be transferred to the other parties so that other parties
will remain in the conference call. Default setting is No"




3. Allow Incoming SIP Messages from SIP Proxy Only

Enabling that should help block SIP scanners/hackers. However, it might prevent Fongo Mobile callers from being able to ring you if you're registered on voip2.freephoneline.ca or voip4.freephoneline.ca:6060
Can someone please test calling from Fongo Mobile to FPL while registered on voip2.freephoneline.ca for me, please, with "Check SIP User ID for Incoming INVITE" set to "yes"?

If that doesn't work, the setting should probably be left at no--and not "yes".

4. Set Check SIP User ID for Incoming INVITE to Yes

This should help block SIP Scanners and hackers.

5. Outgoing Call without Registration

You can leave that set to "no" if you'd like. At least you'll know if you're not registered more easily.
That is, when your ATA is not registered with FPL, outbound calls won't work.

However, strictly speaking, device registration with FPL is not required for outbound calls. It is required for inbound calls only.

6. Subscribe for MWI should be set to No. Freephoneline sends unsolicited MWI.

I don't have time to go through every single setting, but those are the ones that jumped out at me. A quick, cursory glace suggests the rest of them are fine.

Thank you

I have made the last recommended changes and I also made notes on the pdf where SIP user & pass should go.

Thank you for your help


Edit by LiptonBrisk:

A. Change Check SIP User ID for Incoming INVITE to Yes for the reason mentioned previously

B. Subscribe for MWI should be set to No. Freephoneline sends unsolicited MWI

C. Enabling "Allow Incoming SIP Messages from SIP Proxy Only" is a good idea for preventing SIP scanners/hackers. However, you may want to change that setting to "No" because FPL to FPL and Fongo to FPL calls may not work with that setting enabled, depending on the server being used and how switches FPL uses are configured. Fongo Home Phone has this setting disabled.

D. For Banking DTMF (key press recognition), you may need to select "Yes" for " 'Disable DTMF Negotiation": visit viewtopic.php?p=81384#p81384 for full instructions.

E. I suspect since Freephoneline enforces a strict Expires value that must not be less than 60 minutes, that enabling "Unregister on Reboot", which sends Expires=0, won't work properly. I may be mistaken. I note that all other guides for older Grandstream ATA models have "Unregister on Reboot" disabled.

F. Failover server settings can be found here: viewtopic.php?t=20961

G. Please note that no one should be entering an outbound proxy server: viewtopic.php?t=20968

TankCla wrote:I have made the last recommended changes and I also made notes on the pdf where SIP user & pass should go.

Thank you for your help

Thank you, but I am hoping for someone to answer the questions I asked as well because I genuinely do not know the answers to them.
I'm uncertain what the settings should be for #2 and #3 in my previous response.

I would change Check SIP User ID for Incoming INVITE to Yes for the reason I mentioned previously.

Subscribe for MWI should be set to No. Freephoneline sends unsolicited MWI.

I just wanted to say thank you to whoever created and posted these settings. They were very helpful to me in setting up the ht802.

I know this post is old, but I have also a Grandstream device, and I would like to suggest few changes , maybe somebody could use:

1)Since "Prefer Primary SIP Server" is set to NO, maybe it is a good idea to add the "Failover SIP Server=voip2.freephoneline.ca
I know, some may say, if one server goes down, the second one will likely go down too, or not ?

2)Some forum user reported that SIP T1 Timeout 0.5 sec is too agressive, suggesting to set to 1 sec

3)Use # as Redial key: YES
Often when I enter account numbers, credit card numbers, etc....I need to press the number # sign at the end; Redial would create a problem
Better to have as : NO

4)Disable # as Redial key: NO
this sounds like a duplicate line, same a previous, better to have it :YES

5)Ring Frequency: 20
I think it should be 25 HZ, or maybe some phones would not ring

jenom wrote:Since "Prefer Primary SIP Server" is set to NO, maybe it is a good idea to add the "Failover SIP Server=voip2.freephoneline.ca
I know, some may say, if one server goes down, the second one will likely go down too, or not ?

Enabling "Prefer Primary SIP Server" means the account will register to the primary SIP server if registration with Failover SIP server expires.

Specifying Failover SIP Server makes sense. If you do so, enable Prefer Primary SIP Server (set to Yes).

Some forum user reported that SIP T1 Timeout 0.5 sec is too agressive, suggesting to set to 1 sec

Leave at default. It doesn't matter that much because the ATA will will eventually continue to re-transmit until a 2 second (4*T1) timeout is reached.
It also matters less now due to faster ISP services in general.

T1 represents the round-trip time between the ATA and server.

"For example, the HT801/HT802 will attempt to send a request to a SIP
server. The time it takes between sending out the request to the point of getting a
response is the SIP T1 timer. If no response is received the timeout is
increased to (2*T1) and then (4*T1). Request re-transmit retries would
continue until a maximum amount of time defined by T2."

Use # as Redial key: YES
Often when I enter account numbers, credit card numbers, etc....I need to press the number # sign at the end; Redial would create a problem
Better to have as : NO

Disable # as Redial key: NO
this sounds like a duplicate line, same a previous, better to have it :YES

Does the "#" button function as Redial while a call is active? That seems unlikely, unless a bug is present.

Ring Frequency: 20
I think it should be 25 HZ, or maybe some phones would not ring

20 Hz is correct for North America.

Latest firmware versions can be found at https://www.grandstream.com/support/firmware.

For DMTF bank issues, try these steps:


1a) Dial ***
b) Then dial 02
c) Enter the IP address you hear into a web browser.
d) Login to your ATA.

Default login password is admin.


2. Navigate to FXS Port tab used for Freephoneline-->Preferred DTMF method (in listed order)

Priority 1 should be RFC2833
Priority 2 should be In-audio
Priority 3 should be SIP INFO

3. Change "Disable DTMF Negotiation" to "Yes (use above DTMF order without negotiation)"

4. Use First Matching Vocoder in 200OK SDP should be "No"


5. Preferred Vocoder
(in listed order):

choice 1: PCMU

The only codecs that Freephoneline supports are G.711u (PCMU) and G729a (which sounds worse than PCMU).

Don't use G729a for banking or DTMF. I would set all choices to PCMU.


6. Click "Apply" and "Reboot"

---
You can check whether all phone buttons are recognized by calling the free 416 or 250 numbers listed at http://thetestcall.blogspot.com. Wait for voice message. Press # to end echo test.
Press 2 to enter DTMF testing. Press all digits on phone number pad. Press #. What you pressed should be read back to you.
If that works, then all the buttons on your phone are fine, and the current DTMF setting the ATA is using worked for the specific system (not your banking system) being called.

By the way, click viewtopic.php?t=20963

Liptonbrisk wrote: 06/10/2024 Latest firmware versions can be found at https://www.grandstream.com/support/firmware.

For DMTF bank issues, try these steps:


1a) Dial ***
b) Then dial 02
c) Enter the IP address you hear into a web browser.
d) Login to your ATA.

Default login password is admin.


2. Navigate to FXS Port tab used for Freephoneline-->Preferred DTMF method (in listed order)

Priority 1 should be RFC2833
Priority 2 should be In-audio
Priority 3 should be SIP INFO

3. Change "Disable DTMF Negotiation" to "Yes (use above DTMF order without negotiation)"

4. Use First Matching Vocoder in 200OK SDP should be "No"


5. Preferred Vocoder
(in listed order):

choice 1: PCMU

The only codecs that Freephoneline supports are G.711u (PCMU) and G729a (which sounds worse than PCMU).

Don't use G729a for banking or DTMF. I would set all choices to PCMU.


6. Click "Apply" and "Reboot"

---
You can check whether all phone buttons are recognized by calling the free 416 or 250 numbers listed at http://thetestcall.blogspot.com. Wait for voice message. Press # to end echo test.
Press 2 to enter DTMF testing. Press all digits on phone number pad. Press #. What you pressed should be read back to you.
If that works, then all the buttons on your phone are fine, and the current DTMF setting the ATA is using worked for the specific system being called.

By the way, click viewtopic.php?t=20963

My Bank randomly challenges me with a "called" secure login by asking me to enter certain numeric keypads on my phone when I try to login to my accounts. Obviously I cannot accept a "text" secure login attempt because a pots Phone and freephoneline will not work with this, (or will it?).

Anyway, this was not a problem with my older SPA-2102, but with my newer HT802v2 this was a problem because it would fail on a "called" secure login attempt and just drop the line.

But thanks to your DTMF config suggestions above, (for the HT802) I am now able to use the numeric keypads on my (cordless) phones (via my HT802v2) to securely login in ...
Thanks.

tanny wrote: 10/11/2025 Obviously I cannot accept a "text" secure login attempt because a pots Phone and freephoneline will not work with this, (or will it?).

You are right. Freephoneline doesn’t support SMS.

Thanks.

You’re very welcome! I hope you have a great weekend.

Here's how to setup server failover: viewtopic.php?t=20961.

In your Device Configuration Guide, for "Register Expiration" it is set at 60.
Shouldn't it rather be 3600, as per FPL's guideline for "Registration Expires 3600"?
I'm setting up a HT802 V2 with the latest firmware version 1.0.5.9.
Thanks!

Maxwell Smart wrote: 10/16/2025 In your Device Configuration Guide

While that's not my guide, the 3600 value required by FPL is in seconds.

https://support.freephoneline.ca/hc/art ... redentials

3600 seconds is 60 minutes.

for "Register Expiration" it is set at 60.

60 minutes

Shouldn't it rather be 3600, as per FPL's guideline for "Registration Expires 3600"?

The setting in this guide is clearly shown in minutes. So, adjust accordingly.

3600 seconds is 60 minutes. Do not enter 3600 minutes.

Pay close attention to the units used in your ATA's webUI.

https://documentation.grandstream.com/k ... ion-guide/

"Refreshes registration periodically with specified SIP proxy (in minutes). Maximum interval is 65535 minutes (about 45 days).
Default is 60 minutes (or 1 hour)."

Hi Liptonbrisk, Thank you very much! You saved me from making a big mistake. Seconds v.s. minutes. The online setup webpage from FPL was different than the link you kindly shared with me, and the one I had just said 3600 without quantifying it as seconds (or hours or whatever). See attached screenshot. All is good now, Thanks again for helping me

Maxwell Smart wrote: 10/16/2025 See attached screenshot.

Yes, those were originally recommended settings are for older Linksys ATAs (PAP2). Take a look at viewtopic.php?t=20549.
Also, "[6-7]x*xxxxxxxxxxx." doesn't make sense and isn't applicable to anything normal. No one is dialing 6 or 7 followed by another digit, then *, followed by at least 10 more digits.

The old Grandstream HT-286 ATAs also used seconds instead of minutes.

All is good now, Thanks again for helping me

You're welcome!

Thanks again, Liptonbrisk. It works like a charm now.
As for the dial plan setting, rather than the outdated Linksys ATA's string, I'm instead using:
{[2345689]11|*98|[2-9]xxxxxxxxx|1[2-9]xxxxxxxxx|011xxxxxxxxxxxx.}

which seems to work fine. The stock default from the Grandstream factory was: { x+ | \+x+ | *x+ | *xx*x+ }

The Grandstream HT802 V2 has quite a capable web settings GUI. Tons of settings to play around with. Many are well beyond my pay-grade, lol

I also have my Obihai 100, which has worked on FPL super well since I bought it new in 2012. A bit of an antique, but runs rock solid.
Thanks again, much appreciated.

Maxwell Smart wrote: 10/16/2025 I also have my Obihai 100, which has worked on FPL super well since I bought it new in 2012. A bit of an antique, but runs rock solid.

Unless you're faxing (OBi100 does not support T.38 fax protocol, which works with Freephoneline), the OBi100 offers more features that work with Freephoneline than Grandstream ATAs: you can specify all three SIP servers for failover; with Grandstream HT-802 you can only specify two. With Obihai's X_InboundCallRoute you can route inbound calls automatically to other numbers based on the caller ID of the incoming call, you can record calls while a call is active from the call status webUI menu, and you can see useful statistics (dropped packets, call quality, etc.) while on a call by navigating to status-->call status.

Obihai was acquired by Poly, and HP owns Poly. The latest models are the Poly 402 and 400 ATAs, which do support T.38 fax protocol (as do OBi2xx and OBi3xx series ATAs, but they're discontinued).

Thanks again, much appreciated.

You're very welcome!

Yes indeed, Liptonbrisk, the Grandstream only provides for 2 SIP servers for failover, which I implemented thanks to your tips and also to user member TankCla. With my Obi 100, I implemented all 3 SIPS, for failover, which is peace of mind.

I'm setting up the Grandstream HT-802 for my friend who actually still owns a fax machine, so it'll be interesting to see if it works with FPL (more of an experiment, as he normally doesn't use a fax machine any more, I mean, who does?! lol).

I didn't know about the other 2 bonus features of the Obihai. That's pretty nifty (selecting through a caller ID to forward it to another number, and also recording a call). For technology that came out in 2012 or so, it's pretty amazing. Hopefully my Obihai will last for a while longer. Not sure if it has electrolytic capacitors in it that I may have to recap as it ages. I'd probably do that if needed, because it's such a great little ATA.

Maxwell Smart wrote: 10/17/2025 I'm setting up the Grandstream HT-802 for my friend who actually still owns a fax machine, so it'll be interesting to see if it works with FPL (more of an experiment, as he normally doesn't use a fax machine any more, I mean, who does?! lol).

I do, occasionally.

Many physicians, clinics, hospitals, labs, long‑term care homes and services, and pharmacies still send referrals, requisitions, consult notes, and prescriptions by fax. Some insurance brokers, lawyers, and real estate agents also fax and request faxes. Fax persists where systems are fragmented, legal proof of transmission is valued, and interoperability standards are still maturing. Provinces and the federal government are pushing standards and e‑services, but the shift away from fax is incremental.

Anyway, visit viewtopic.php?p=77863#p77863 for fax settings that may help your friend.

I'd probably do that if needed, because it's such a great little ATA.

Obihai ATAs (and now Poly 4xx series ATAs) offer the most features that can be used with Freephoneline. You can also create whitelists, blacklists, and force all other callers to press "1" to ring through.