Error: Outgoing calls always sound occupied

[f_laplante]

ATA is registered - Grandstream HT-502
Incoming calls Work fine

Outgoing calls Always Sound Occupied. A new issue since…?

I’ve been using freephoneline for more than 10 years. I have paid the new 911 fees To keep the service alive. Is there anything that has changed recently?
Thx

[Liptonbrisk]

What does “sound occupied” mean?

I have no issue with outgoing calls with multiple FPL numbers on multiple devices.

[f_laplante]

It sounds like this
https://youtu.be/sMN_wAvGVng?si=GNU4IKn9oYIkYFQu

[Liptonbrisk]

You posted in the Fongo Home Phone forum originally. I'll move this thread to the Freephoneline forum.


Okay, a busy signal
(If you hear a stutter tone before dialing a number, press *98 to check to for voicemail.)


Busy signals from outbound calls can be due to incorrect dial plans in the ATA, not dialing the right number (unless this is happening on all outbound calls), a faulty handset/DTMF issue (pressed button isn't working), NAT firewalls blocking ports (but this is less likely if inbound calls are working), or SIP ALG being enabled in the router.

Please provide information from or follow the steps from 1,2,4, and 6: viewtopic.php?t=20199. I need that information.


1) What brand and model modem are you using? (I'm not sure if you're still using a Thompson cable modem, but I need the model regardless)

2) What brand and model router are you using? (I'm not sure if you're still using an Apple Airport extreme).

3) Ensure SIP ALG is disabled in whatever router you're using (or the modem/router combo issued by your ISP). Contact your ISP if necessary for assistance.

4) What proxy server are you using in the ATA (voip.freephoneline.ca, voip2.freephoneline.ca, or voip4.freephoneline.ca:6060)?

6) Please also login at https://www.freephoneline.ca/callLogs and check to confirm that the "To" and "From" fields represent the full phone numbers you expect. You want to make sure you dialed the number properly. Double check the "to" field.

If you're using a phone handset that has a call log, does it show the full and proper phone number has been entered properly?

If you never touched the dial plan in the ATA, and if the existing dial plan as always worked for you with Freephoneline, then the dialplan probably isn't causing the problem. In that case, it's more likely to be a faulty handset (button no longer working properly), incorrectly dialed number, DTMF issue, SIP ALG, or firewall problem.

You can check your DTMF settings using this guide: viewtopic.php?p=64438#p64438. Also take a look at viewtopic.php?p=81016#p81016 if you encounter TD banking issues. DTMF settings affect whether keypresses are recognized properly.

Another possibility is a codec mismatch (but because inbound calls are working, I suspect this isn't the cause). Freephoneline only supports G.711U and G.729a audio codecs. If you never changed audio codec settings in the ATA, then I doubt it's an audio codec mismatch problem.


You can also try step 9 from viewtopic.php?t=20199. If that reboot sequence works, you were likely dealing with NAT corruption.

[f_laplante]

Hey! lots of information... a lot more than 10 years ago!

My info are :
CABLE TO modem ARRIS TM822G LAN
TO WAN Linksys WRT3200ACM LAN 1
TO WAN ATA Grandstream ht502 Fx1 freephoneline Fx2 freephoneline

I was not able to check for SIP at this point.

I'm using voip.freephoneline.ca, voip2.freephoneline.ca,

When looking to my call log, I see a strange message :

0:00 Mar 22, 2025 10:46:11 PM Mar 22, 2025 10:46:11 PM Valid cause code not yet received St-Lambert $ 0.00

What does it means, "Valid cause code not yet received"??? Can it explain my problem???

[Liptonbrisk]

While I'm looking up information on your equipment, please check to see whether you can call 416-342-9562 from https://thetestcall.blogspot.com/.
If you can call that Toronto number, wait for the beep, press #, and then press 2. Then press every single digit on your phone. Then press #. Wait to hear if each button was successfully pressed. Then end the call. That way you can test to ensure DTMF is working properly.

If you can call 416-342-9562 successfully, submit a ticket: https://support.fongo.com/hc/requests/new
Only submit a ticket if you can call other numbers fine, such as the Toronto number I listed above. I can call 416-342-9562. It works.
Choose Freephoneline-->My Account Inquiry for the final issue type. In the ticket, provide the full St-Lambert phone number that you're having problems calling along with the entire call log entry for the failed call. Mention that you are able to call other numbers and that the problem appears to be a carrier issue.

Afterwards, private message me with the ticket number: ucp.php?i=pm&mode=compose&u=1229. Again, only do this if you can call 416-342-9562 successfully.

--
(General info)

Visit https://status.fongo.com/.
If "Support System" indicates "Partial Outage" or "Degraded Performance", ticket response time can take up to a week (or longer).


You can check your ticket status by logging in at
https://support.fongo.com/hc/requests. That's an account for tickets
(zendesk) only and is completely separate from your Freephoneline account or
any other Fongo account you may have. If you don't have a zendesk account yet, click "Sign Up" after visiting the link.
Use the same email address that you use to submit tickets. Do not use the same password as your Freephoneline or Fongo Mobile
account. Again, these accounts are unrelated.




Support staff does not respond to tickets on weekends or Canadian holidays.
Support hours are 9 a.m. until 4 p.m. EST.

Fongo does have https://twitter.com/Fongo_Support. I'm not sure if anyone
there responds to direct messages.

Similarly, they appear to be on Facebook:
https://www.facebook.com/FongoMobile/. I don't know whether they'll respond to you there.

[Liptonbrisk]

CABLE TO modem ARRIS TM822G LAN

Okay that's not a modem/router combo or hub. It doesn't have router features.

TO WAN Linksys WRT3200ACM LAN 1

I don't see a way to disable SIP ALG using official Linksys firmware.

https://downloads.linksys.com/downloads ... tional.pdf
I'm pretty sure if the option to disable SIP ALG were available, it would be shown on page 18. This doesn't necessarily mean SIP ALG isn't on. The option to disable it could simply not be available. It depends on router firmware.

I'm using voip.freephoneline.ca, voip2.freephoneline.ca

If you can't call the Toronto number I provided earlier, then (follow the steps below, step by step down the list) try switching to "voip4.freephoneline.ca:6060" without the quotation marks.

1.
a) 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. For HT-503, navigate to the FXS Port tab.

A. If you could be dealing with a SIP ALG problem (ISP's hub or gateway that’s been provided has SIP ALG on with no way for the user to disable it, or if you can't figure out how to disable SIP ALG in your own router), navigate to “Primary SIP Server”.


Enter "voip4.freephoneline.ca:6060" (without the quotation marks) to help avoid any potential SIP ALG bug. You should be testing with voip4.freephoneline.ca:6060 if you're getting 1-way audio problems (one side hears audio, and the other side doesn't), or if calls are failing. Anyone can use voip4.freephoneline.ca:6060, even people who don't use Rogers. If you have 1-way audio problems, use it.


B. “NAT Traversal” should be set to “Keep-Alive” (newer ATA models) or “Yes” with the STUN server field blank (Grandstream HT-286/HT-287). Don't use STUN (when the STUN server drops so will your Freephoneline service).

C. Enable SIP OPTIONS/NOTIFY Keep Alive

Select “Notify”



D. Set SIP OPTIONS/NOTIFY Keep Alive Interval to 20 seconds.



E. Register Expiration needs to be 3600 seconds or 60 minutes.

Please take note of whether your ATA’s firmware designates minutes or seconds for this setting.

The value needs to be either 60 minutes or 3600 seconds (1 hour).
You can’t enter 60 seconds without getting your Freephoneline account blocked. Be careful here.

HT-503/701/704/8xx series ATAs appear to use minutes for this setting. If so, the correct value is 60 minutes. Double check.

F. i) Set “Use Random SIP Port” to “Yes”.
ii) Set “Random RTP Port” to “Yes”.


G. Set “SIP Registration Failure Retry Wait Time” to 120 seconds


H. Refer to page 4 (bottom) of the PDF Guide located at viewtopic.php?p=64438#p64438. The PDF guide link is located at the bottom of the first post.

a) The preferred Vocoder (in listed order) should be PCMU (first choice), which is the G.711u audio codec. G729 can be the second choice, but it sounds worse. If you have any other codec listed, change it to PCMU. Freephoneline only supports PCMU (G.711u) and G.729a. If you use anything else, calls can fail.


3. Update/Apply (save) ATA settings if changes were made. Turn off modem, router, and ATA. Turn on modem. Wait for it to be fully up and running first. Turn on router. Wait for router to be fully up and transmitting data first. Lastly, turn on ATA after everything else is up and running. That's always the proper device boot order. Your ATA should always be booted last in the device chain. 1. Modem (wait) -->2. Router (wait)-->3. ATA

4. Login at https://www.freephoneline.ca/showSipSettings.
SIP Status needs to indicate "Connected", and SIP User Agent should reflect the device you're using.

5. Login to your ATA. Ensure your FPL account is registered with your ATA.

For HT-503/701/704/8xx series ATAs, navigate to Status–>Port Status–>FXS–>Registration (also ensure DND shows “No”).

[Note that unless you own two VoIP unlock keys, you can't register two Grandstream FXS ports with your FPL account simultaneously.

When there are multiple devices/softphones/FXS ports (on an ATA) using the same FPL account and SIP server, only the most recent registration is valid. The previously registered device (or FXS port, in this case) will lose registration, and, consequently, incoming calls will not ring on it. This is especially important to consider if someone else is using your SIP credentials (username and password) that are found after logging in at https://www.freephoneline.ca/showSipSettings or if you're trying to register your FPL account with a smartphone SIP app (or FPL desktop app), with another device, or more than one ATA line. Registration is required for incoming calls. It is not required for outgoing calls. Only one registration per FPL account is allowed at any time. A single FXS Port on a Grandstream ATA counts as one registration. A SIP app is another.]


6. Try calling the Toronto number at this point. If it doesn't work, check the dial plan in the ATA.

Try using this:

Code: Select all

 {[2345689]11|*98|[2-9]xxxxxxxxx|1[2-9]xxxxxxxxx|011xxxxxxxxxxxx.}

Official VoIP Unlock Key settings are found at https://support.freephoneline.ca/hc/en- ... redentials.

When looking to my call log, I see a strange message :

0:00 Mar 22, 2025 10:46:11 PM Mar 22, 2025 10:46:11 PM Valid cause code not yet received St-Lambert $ 0.00

What does it means, "Valid cause code not yet received"??? Can it explain my problem???

It means a response for the reason the call failed wasn't received. Usually there's a response or error code. Possibly the cause of the failed call is due to a carrier issue. I'm not sure. The message suggests to me that you should be submitting a ticket as described in my previous post if calls to other cities work fine.

[Liptonbrisk]

Also, manually type your SIP password shown after logging in at https://www.freephoneline.ca/showSipSettings into the "Authenticate Password" field in your ATA's webUI (under the FXS port tab). Disable password and form autofillers (browser app extensions) that may be filling in the wrong password automatically. You may hear a busy signal on outbound calls in the SIP password hasn't been entered correctly. Do not copy and paste the password.

Even a single incorrect character (e.g., O vs 0, l vs 1), incorrect case (passwords are case-sensitive), or an accidental leading/trailing space can cause failure. If the password was copied and pasted, sometimes hidden characters or formatting can be included. Ensure it's the SIP Password provided by Freephoneline, not the password used to log into the Freephoneline web portal.

ATA will show registered even if the Authenticate Password is entered incorrectly, and if it is, outbound calls won't work. Incoming calls will work if the ATA registered properly within the past hour--before the Authenticate Password was possibly changed incorrectly.