For Grandstream ATAs with incoming call issues/1-way audio

[Liptonbrisk]

A. Always visit https://status.fongo.com/ first to check for reported issues.

B. Please read #20 down below and also understand that NAT corruption can develop between a router and ATA without a user doing anything (no matter how long FPL has been working for someone).

C. Check all cables and cords to ensure they're all secure (try a different phone as well).

D. Avoid using STUN. Using STUN introduces an additional point of failure. When the STUN server goes down, so does your FPL service. Do not specify a STUN server for NAT Traversal in your ATA's web user interface.

E. Device registration is a requirement for incoming calls but not for outgoing ones. Only one registration is permitted per FPL account at any time. A single line (FXS) on an ATA is one registration. A SIP app is another.

F. For Asus router users following the steps below, first login to your router’s web UI.
Navigate to Advanced Settings–>Administration–>System (tab)–>Basic Config–>
Change “Enable WAN down browser redirect notice” to "No".
Click “Apply”. That fixes potential problems with an ATA attempting to register with 10.0.0.1
when it's booted before the ISP's modem is fully up and running first (after a power outage, for example).



Follow the steps, step by step, down the list, please. Some ATA models may have menu locations that differ. Some options may require updating ATA firmware versions.

1. If you're using a modem/router combo, gateway, or hub issued by your ISP (and are NOT using your own additional separate router), contact your ISP to ask for assistance for disabling SIP ALG in the modem/router combo, gateway, or hub. Disable SIP ALG. Refer to point 1 from viewtopic.php?f=8&t=20199.

To understand why SIP ALG often causes horrible problems, please visit
https://www.voip-info.org/routers-sip-alg/ (scroll down to the section on SIP ALG problems).


2. Make sure whatever modem/router combo, gateway, or hub your ISP gave you is in bridge mode if (and only if) you are using your own separate router as well. Call/contact your ISP if you have to.

For Bell and Virgin Hubs, I find it's often simpler to perform PPPoE login in your own router (this is PPPoE Passthrough) and disable Wi-Fi in the hub. You will need the PPPoE Username and Password from Bell or Virgin.

For Rogers, visit https://www.rogers.com/customer/support ... ridgemodem.
Shaw users will have to call Shaw to enable bridge mode at the time of this post.

For Telus Wi-Fi Hubs, put LAN port 1 on the Hub in bridge mode. Then connect your router to LAN port 1.
Alternatively, contact Telus and ask them how to enable bridge mode.


3. If you're using your own router in addition to the gateway or hub provided by your ISP, ensure SIP ALG or SIP Passthrough (Asus routers) is disabled in your own router. Refer to your router's manual.

4. Disable DMZ and all port forwarding in your router. Port forwarding is a security risk. Only port forward if you have no other choice.

5. a) If you're using an Ubiquiti router, disable jumbo frames.

b) This may affect pfSense users (and some others), depending on configuration: don't block incoming UDP connections from 208.85.218.146 and 208.85.218.147 if you want to hear audio. At the time of this post, those are the RTP IP addresses. Those IPs may eventually change.

6. If your ATA is connected to the internet using a third party VPN service, disable the VPN while troubleshooting.

7. Login at https://www.freephoneline.ca/voicemailSettings
Ensure "Rings Before Voicemail" is greater than 1.

8. While troubleshooting incoming call issues, disable Follow Me: login at https://www.freephoneline.ca/followMeSettings.

9. Freephoneline doesn’t support ATAs’ call forwarding features. Incoming calls will drop to FPL’s voicemail system immediately once call forwarding is attempted by the ATA. It’s possible to enable call forwarding by accidentally dialing star codes.

a) To cancel Unconditional Call Forward, dial *73, wait for dial tone, and then hang up.
b) To cancel Busy Call Forward, dial *91, wait for dial tone, and then hang up.
c) To cancel Delayed Call Forward, dial *93, wait for dial tone, and then hang up.

10. Dial *79 to disable Do Not Disturb. It’s possible to accidentally dial a star code and enable Do Not Disturb (all incoming calls will drop directly to FPL’s voicemail system when DND is enabled). I don’t know whether HT-286/HT-287 has the *79 star code option.

Also check your phone handset to ensure Do Not Disturb isn’t enabled in it.

11. 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.


12. The setting locations may differ depending on the firmware version used (the locations mentioned may not be accurate; double check).

For HT-286/HT-287 series, navigate to the Advanced Profile tab used for FPL.
For HT-503, navigate to the FXS Port tab.
For HT-701/HT-702 (firmware 1.0.8.2 or newer) and HT-8xx, navigate to the FXS Port used for FPL.


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”.

For HT-286/HT-287s, the setting is called “SIP server”.

Enter "voip4.freephoneline.ca:6060" (without the quotation marks) to help avoid any potential SIP ALG bug (even if SIP Passthrough is disabled in stock Asus firmware, try voip4.freephoneline.ca:6060 if you're experiencing issues). 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). Anyone can use voip4.freephoneline.ca:6060, even people who don't use Rogers. If you have 1-way audio problems, use it.


(Also remember to enter your SIP UserID, Authenticate Password, and anything else from the PDF configuration guides on the Fongo forums; I’m just emphasizing important settings. The guides are located at viewforum.php?f=15.)


B. “NAT Traversal” should be set to “Keep-Alive” (newer ATA models) or “Yes” with the STUN server field blank (HT-286/HT-287)

C. Enable SIP OPTIONS/NOTIFY Keep Alive

Select “Notify”

HT-286/HT-287 doesn’t appear to have this setting.


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

HT-286/HT-287 setting is called “keep-alive interval” and should also be set 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-286/HT-287 ATAs appear to use seconds for this setting. If so, the correct value is 3600 seconds.

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

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

HT-286/HT-287 users should select “Yes” for “Use random port”.


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


H. Set “SIP REGISTER Contact Header Uses” to “WAN address”

This helps to ensure data is sent back to your public IP address as opposed to
your LAN IP address (192.100.1.x, for example). If Freephoneline were to send data to 192.100.1.x, it would never reach you. It needs to be sent to your WAN or public IP address first before your router can send or route the data to the local IP address of your Grandstream ATA.

Enabling this setting helps to ensure one-way audio issues don't occur.

I don’t believe this setting is available for HT-286 or HT-503.

HT-7xx users may need to update firmware before this setting is available.


I. Disable "Allow Incoming SIP Messages from SIP Proxy Only". Change that to "No". 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


13. Update/Apply 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

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

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

For HT-286/HT-287, navigate to Status–>Registered
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.


16. Test incoming calls, preferably with a traditional landline or regular (non-VoIP) cellular number so that you don't have to spend time troubleshooting the other side of the call too. If incoming calls work at this point, you can stop at this step, but I suggest reading step 20 regardless.


17. If incoming calls are not working, login at https://www.freephoneline.ca/callLogs to confirm whether incoming calls are reaching your FPL account (they should be if incoming calls are reaching your FPL voicemail message). Only calls that are answered in some manner (including being answered by FPL's voicemail system) will appear in FPL's call list. Calls that aren't answered aren't listed. Duration is rounded to the next minute. Check the disconnect reason.

Incoming calls must reach FPL first in order for them to also reach your ATA.


18. If incoming calls are not reaching your ATA but are reaching FPL’s voicemail system, attach the ATA directly to the ISP's (gateway or hub in bridge mode) modem via ethernet cable. Test briefly for incoming calls. If incoming calls suddenly start working properly, you've narrowed down the problem to something involving your router. Keep in mind your ATA will not be protected by your router's firewall during this step. After testing, revert back; that is, ensure your ATA is protected by a firewall again.

[Liptonbrisk]

19. If you can't ping voip.freephoneline.ca, voip2.freephoneline.ca, or voip4.freephoneline.ca when connected to your ISP's hub, gateway, or modem, that may indicate a DNS problem with your ISP.

The simplest way to check for a DNS problem is to replace “voip4.freephoneline.ca:6060” with “163.213.111.21:6060” (without the quotation marks) for Primary SIP Server (or "SIP Server" for HT-286) in step 12A above. If your FXS Port for FPL suddenly registers afterwards, you’re dealing with a DNS problem with the ATA. However, if the IP address for voip4.freephoneline.ca ever changes, FPL will stop working for you again.

You could try using https://www.quad9.net/ or any alternate DNS servers you want in your ATA. You may want to double check your DNS entries in your ATA, regardless, if FPL still isn’t registered at this point.

Ensure you can ping or reach FPL's SIP servers. If you can't, you (or your ATA) may be experiencing a DNS issue.

---
"Test pings and jitter (you want little to no variation between pings) to the specific Freephoneline SIP servers you plan on using.

Use winmtr: https://sourceforge.net/projects/winmtr/. Ping about 200 times to each server.

My pings to
-voip.freephoneline.ca average 11 ms.
-voip2.freephoneline.ca average 12 ms
-voip4.freephoneline.ca average 27 ms

If you're using a Macintosh, maybe this helps: https://www.reddit.com/r/TagPro/comment ... tr_on_mac/

When using WinMTR, look at the very last hop or line. Look at your average ping and then maximum ping. Although WINMTR doesn't provide a jitter value, you can get an idea of what yours is by subtracting maximum ping from your average. Jitter is the difference between each successive ping. The bigger the difference, the bigger the problem.

Same with ping, which represents lag or delay. The lower your ping and jitter, the better.

You do not want high pings and lots of jitter (you do not want a lot of variation between each ping). If you get horrible results (pings over 200ms), to any server, you probably don’t want to use that server. So you would want to give that server the lowest priority.

I get between 11 (voip.freephoneline.ca and voip2.freephoneline.ca) and 24ms (voip4.freephonline.ca) on average, depending on the server I'm testing to. Preferably, you want pings below 100ms.
Anything over 200ms is unacceptable.
What you don't want to see is 40, 45, 50, 35, 500, 40, 30, 45, 700. That's bad jitter.
You want relatively consistent pings without a lot of variation.

One reason why jitter can occur is due to other devices on your LAN (local area network) using bandwidth. That’s why properly enabling QoS in your router for your ATA is always a good idea. Refer to point C from viewtopic.php?f=8&t=20199#p78976.

Bad jitter can produce broken-up audio or choppiness during phone calls. Severe jitter (or large ping spikes) can cause calls to drop (and incoming calls won’t arrive while the ping spike is occurring). Ping affects delay.

I recommend testing pings/jitter between 8 p.m. and 11 p.m. to see if local congestion is a factor (this often is your ISP's fault). Sundays are the best days to test (because that's when most people in your area will be home). 8 p.m. - 11 p.m. is prime time. During prime time (between 8 p.m. and 11 p.m.) cable internet nodes may be oversubscribed in your area and face congestion issues (and congestion can also exist with DSL). So I suggest testing services between 8 p.m. and 11 p.m., particularly on Sundays, when everyone in your area will be home.


20. Thanks to Mango, many of us now understand that in order for ATAs to remain registered and working properly with a VoIP SIP provider like Freephoneline, in particular after power failures, the following conditions must be met:

UDP Unreplied Timeout (in your router) < SIP Notify Keep Alive Interval (in your ATA) < UDP Assured Timeout (in your router) < SIP Registration Failure Retry Wait Time (in your ATA)

“<“ means less than.

When a modem leases a new IP address, a problem can arise where prior associations using the old IP address are maintained in the router. When the ATA attempts to communicate using the old IP address, the response is unreplied, and then if the UDP Unreplied timeout is greater than the Keep Alive Interval (and UDP Unreplied timeout is often set to 30 by default in consumer routers) a problem arises where the corrupted connection persists. If UDP Unreplied timeout is, for example, 15, and the NAT Keep Alive Interval is 20, then the corrupted connection will timeout or close. A new connection will be created, and everything will work fine.

Another problem can occur when the Keep-Alive interval is greater than UDP Assured Timeout (often 180 by default in consumer routers): the NAT hole will close due to the ATA not communicating frequently enough with the SIP server. In turn, incoming calls may, intermittently, not reach the ATA. Again, SIP Notify Keep Alive Interval is supposed to be 20 seconds with FPL.



Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers may be difficult, if not impossible. Asuswrt-Merlin (I would avoid any model below/less powerful than an RT-AC68U), third party firmware for Asus routers, does offer easy access to these two settings, which are found under General–>Tools-->Other settings. My understanding is that third party Tomato firmware has these two settings as well. So if your router supports Tomato firmware, that may be another option. Note that I will not be held accountable any damage resulting from failed firmware updates. Apparently, Mikrotik routers also allow users to change both Assured and Unreplied UDP timeout settings as well: https://forums.redflagdeals.com/recomme ... #p28059363.

Router firmware that allows users to adjust Assured and Unreplied UDP timeouts include

Asuswrt-Merlin
Ubiquiti
Mikrotik
pfSense
Tomato
DD-WRT
OpenWrt


The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 15 for UDP Unreplied Timeout and 115 for UDP Assured Timeout.

[Liptonbrisk]

(continuing point 20)


a. Asuswrt-Merlin

i) Login to router's web UI
ii) Navigate to General-->Tools-->Other Settings
iii) Change "UDP Timeout: Assured" to 115 seconds if the failed registration retry timer in your ATA or IP Phone is 120 seconds for Freephoneline.
iv) Change "UDP Timeout: Unreplied" to 15 seconds if the NAT Keep-alive Interval in your ATA or IP Phone is 20 seconds for Freephoneline.
v) Click "Apply"


b. Ubiquiti

i) Login to Unifi Controller
ii) Navigate to Routing & Firewall-->Firewall-->Settings-->State Timeouts
iii) Change "UDP Stream" to 115 seconds if the failed registration retry timer in your ATA or IP Phone is 120 seconds for Freephoneline.
iv) Change "UDP Other" to 15 seconds if the NAT Keep-alive Interval in your ATA or IP Phone is 20 seconds for Freephoneline.
v) Click "Apply Settings" (Save changes made).


c. Mikrotik

i) Use Winbox: https://download2.mikrotik.com/routeros ... winbox.exe
To learn how to connect to your router, visit https://wiki.mikrotik.com/wiki/Manual:Winbox. Connect to your router and login.
ii) Enter "ip firewall connection tracking set udp-stream-timeout=115s" (without the quotation marks) if the failed registration retry timer in your ATA or IP Phone is 120 seconds for Freephoneline.
iii) Enter "ip firewall connection tracking set udp-timeout=15s" (without the quotation marks) if the NAT Keep-alive Interval in your ATA or IP Phone is 20 seconds for Freephoneline.

d. pfSense

UDP Multiple is UDP Assured
UDP Single is UDP Unreplied
Based on https://www.netgate.com/docs/pfsense/bo ... l-nat.html

i) Login to pfSense GUI.
ii) Navigate to System-->Advanced-->Firewall & NAT-->Firewall Optimization Options
Scroll down to "State Timeouts".
iii) Change UDP First (udp.first) to 115 seconds if the failed registration retry timer in your ATA or IP Phone in your ATA or IP Phone is 120 seconds for Freephoneline.
iv) Change UDP Single (udp.single) to 15 seconds if the NAT Keep-alive Interval in your ATA or IP Phone is 20 seconds for Freephoneline.
iv) Change UDP Multiple (udp.multiple) to 115 seconds if the failed registration retry timer in your ATA or IP Phone in your ATA or IP Phone is 120 seconds for Freephoneline.
v) Save settings


e. Tomato

i) Login to router's web UI
ii) Navigate to Avanced-->Conntrack / Netfilter-->UDP Timeout
iii) Change "UDP Timeout: Assured" to 115 seconds if the failed registration retry timer in your ATA or IP Phone is 120 seconds for Freephoneline.
iv) Change "UDP Timeout: Unreplied" to 15 seconds if the NAT Keep-alive Interval in your ATA or IP Phone is 20 seconds for Freephoneline.
v) Click "Save"

f. DD-WRT (I've never used DD-WRT and am not able to test whether this works)

i) Login to router web UI.
ii) Navigate to Administration-->Commands (use command shell). Or SSH/Telnet into your router.

Enter the following:
iii) "echo 115 > /proc/sys/net/ipv4/netfilter/ip_conntrack_udp_timeout_stream" (without the quotation marks) if the failed registration retry timer in your ATA or IP Phone is 120 seconds for Freephoneline.
iv) "echo 15 > /proc/sys/net/ipv4/netfilter/ip_conntrack_udp_timeout" (without the quotation marks) if the NAT Keep-alive Interval in your ATA or IP Phone is 20 seconds for Freephoneline.

Note: It's possible these changes may not be saved in DD-WRT after rebooting.
https://www.linksysinfo.org/index.php?t ... ost-274528

To persist changes after reboot, you need to add your command to crontab or "startup scripts".
In my case I had to shove the damn assignment into crontab, because either the startup command fails sometimes or the value gets reset eventually

One of these two UDP settings is adjustable in DD-WRT web UI at Administration-->Management-->IP Filter Settings-->UDP Timeout (in seconds), but depending on the firmware version used, the single UDP timeout setting that is adjustable differs.


g. OpenWrt

Add the following (or change) to /etc/sysctl.conf


i) net.netfilter.nf_conntrack_udp_timeout_stream=115 if the failed registration retry timer in your ATA or IP Phone is 120 seconds for Freephoneline.
ii) net.netfilter.nf_conntrack_udp_timeout=15 if the NAT Keep-alive Interval in your ATA or IP Phone is 20 seconds for Freephoneline.

Then run sysctl -p to load the new settings from the file.