Fongo Community Forums

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.

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 being called.