Always visit https://status.fongo.com/ first to check for reported issues.
Please read #28 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).
Check all cables and cords to ensure they're all secure (try a different phone as well).
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 on an ATA is one registration. A SIP app is another.
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 slightly, but they should be very similar.
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. a) Dial ****
b) Then dial 110#
c) Enter the IP address you hear into a web browser.
d) Login to your ATA.
e) Always choose the admin login and advanced view menus (select "advanced" in the upper right).
For Linksys RT31P2, try logging in at http://192.168.15.1/Voice_adminPage.htm
Or try adding "/Voice_adminPage.htm" (without the quotation marks) after the Linksys RT31P2's LAN IP in a web browser.
10. A. Under User 1 and User 2 tabs
B. Select Advanced View
C. Under Supplementary Service Settings
Ensure
a) DND setting is set to NO
b) Block ANC Setting is set to NO
c) DND Activated is set to NO
D. Under User 1 and User 2 tabs (both of these, if your ATA offers both tabs--and if you're using both with Freephoneline; otherwise just choose the tab you're using with FPL)
Check Call Forward Settings
a) Cfwd All Dest
That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.
b) Cfwd Busy Dest
That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.
c) Cfwd No Ans Dest
That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.
Submit all changes/Save Settings if changes were made.
11. If you have an SPA3102 (or an ATA with a PSTN line option) and don't have a traditional telephone landline service (aren't using PSTN), then
a) navigate to Voice tab-->Line 1 tab-->VoIP Fallback To PSTN, and set "Auto PSTN Fallback:" to "No".
Click "Submit All Changes" if changes were made.
and
b) navigate to Voice tab-->PSTN Line tab-->set Line enable to "No"
Click "Submit all Changes" if changes were made.
Note that the "Phone" port on the back of the SPA3102 is for calls made using Line 1. The "Line" port on the back of the ATA is for the PSTN Line (connecting to a traditional telephone service).
12. Navigate to Voice tab-->Line 1 tab (or whichever Line you're using for FPL)-->SIP settings, and change (local) SIP Port to a random number between 30000 and 60000. Specify a high random (UDP) SIP port in your ATA between 30000 and 60000. Just choose a UDP port number in that range. If you already have a random number in that range entered, choose a new random number within the same range, and enter it.
Do not use the same random SIP port for any other Line. Always choose a different random local SIP port for each Line you're using. Never use UDP port 5060.
Using a high random SIP port may help to bypass SIP ALG, and it also helps to avoid SIP Scanners (or hackers).
Also, changing local SIP port will reset a potential corrupted NAT association that developed between your router and ATA due to a UDP timeout issue (if a similar problem arises in the future, repeat this step, and change to a different port number within the same range).
Save settings/Submit all settings.
Reboot ATA.
Test incoming calls at this point. I just want to check at this point to see whether changing X_UserAgentPort helped. If it did, the problem was NAT corruption or a UDP timeout (refer to point 28 down below) related issue in your router, which changing X_UserAgentPort helped to reset.
13. Log back into your ATA using a web browser. If you could be dealing with a SIP ALG problem (ISP's hub or gateway has SIP ALG on with no way for user to disable it or if you can't figure out how to disable SIP ALG in your own router), navigate to the Line used for FPL-->Proxy and Registration->-Proxy. Use "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 Username, SIP Password, and anything else from the PDF configuration guides on the Fongo forums; I’m just emphasizing important settings). They are located at viewforum.php?f=15.
14. Navigate to Voice tab-->Line tab (whichever you use for FPL)-->NAT settings
a) NAT Mapping Enable should be yes
b) NAT Keep Alive Enable should be yes
c) NAT Keep Alive Msg should be $NOTIFY
d) click "Save Settings" button if changes were made
15. In your ATA, navigate to Voice tab-->Line tab (whichever you use for FPL)-->Proxy and Registration-->Register Expires needs to be 3600 seconds
save settings
16. In your ATA, navigate to Voice tab-->Line tab (whichever you use for FPL)-->Supplementary Service Subscription-->Block CID Serv:
a) change to no
b) Click "submit"
Do this to avoid 15 minute call disconnections with Freephoneline.
17. Navigate to Voice tab-->SIP tab-->NAT Support Parameters, and make sure that the following settings are enabled:
a)Handle VIA received-->yes
b)Handle VIA rport-->yes
c)Substitute VIA Addr-->yes
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 data to your ATA's local IP address.
Enabling this setting helps to ensure one-way audio issues don't occur.
d) NAT Keep Alive Interval should be 20 seconds
e) Stun enable: No
f) Stun Test enable: No
g) Delete the STUN Server field if anything is listed
Using STUN creates an additional point of failure. When the STUN server goes down, so does your FPL service.
h) click "Save Settings" button
18. Navigate to Voice tab-->SIP tab-->SIP Timer Values (sec)
Reg Retry Intvl needs to be 120 seconds at least.
Click "Save Settings" button if changes were made
Many older guides for FPL don't include this setting.
19. Save/submit settings. 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. ATA should always be booted last in the chain. 1. Modem (wait) -->2. Router (wait)-->3. ATA