FPL no longer Working

[robchiappetta]

I am having issues with my FPL service. I am with Cannet Tel ISP, and have a Linksys EA9500 router with a Linksys PAP2T. I have the PAP2T configured with voip.freephoneline.ca and my username and password but when I try and make a phone call, I get a busy signal everytime, no matter which number I call. When I call into my line, it goes straight to voicemail.

I have also tried voip2.freephoneline.ca and voip4.freephoneline.ca:6060 but the same issues occurs. I have also ensured I have UDP ports open 5060, 5061, 6060, 6061, 13000 and 13001.

Any help would be appreciated

Thank you

[Liptonbrisk]

The next time this happens, as a very first step, please always check the registration state both in your ATA (Info-->Line (FPL) status)and after logging in at https://www.freephoneline.ca/showSipSettings.
Your ATA needs to be registered; SIP status needs to be connected; and if you see a SIP User Agent in your FPL account portal that you don't recognize, you've been hacked.
You should also check call logs at https://www.freephoneline.ca/doGetCallLogs to see if your calls are actually completing.

Afterwards, visit http://status.fongo.com/ to check server status.
If the service status website doesn’t note any issues, then chances are the problem is on your end.

I am with Cannet Tel ISP

Please follow the steps below from 1 to 12, slowly and carefully, step by step in the order they are listed.

1. If your ISP gave you a modem/router combo or gateway, ensure that it's in bridge mode so that you bypass that device's NAT firewall and potentially buggy SIP ALG feature. Refer to the device manual, or contact your ISP if need be. If they gave you a DSL modem/router combo or gateway, perform PPPoE login using your Linksys router.


Connect your ATA to your Linksys router.

and have a Linksys EA9500 router

2. Disable SIP ALG in it.
Login to your router. Navigate to Router Settings-->Connectivity-->Administration-->Application Layer Gateway
Uncheck "SIP". Click "Apply" and "Ok".

with a Linksys PAP2T.

3. For reference, your PDF setup guide can be found here: http://forum.fongo.com/viewtopic.php?f= ... 294#p64346

These settings should be set as followed:

a) NAT Mapping Enable: yes
b) NAT Keep Alive Enable: yes
c) NAT Keep Alive Intvl: 20

Save settings


4. In your PAP2T, Navigate to Line 1 (or whatever you're using for FPL)-->SIP settings, change SIP Port to a random number between 30000 and 60000. Just pick a random number in that range.
Save settings.


If changing the local SIP Port fixes a failed registration state, you were dealing with a corrupted NAT connection in your router.
Possibly a NAT router connection was never disconnected or never timed out properly. And, then, the
ATA keeps the corrupted connection in a persistent state over and over again. (Credit goes to Mango for
this information). Possibly, this problem is due to the router's UDP timeout being in excess of the ATA's
Failure Retry timer. With FPL, that's 120 seconds.

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) < NAT Keep-alive Interval < UDP Assured Timeout (in your router) < SIP Registration Failure Retry Wait Time (Reg Retry Intvl)

“<“ 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, 10, 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, NAT Keep Alive Intvl is supposed to be 20 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, 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.
The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time (Reg Retry Intvl setting in your ATA) is 120. I use 10 for UDP
Unreplied Timeout and 117 for UDP Assured Timeout.


5. In your PAP2T, Navigate to the 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

Save settings

6. Navigate to Voice-->SIP-->SIP Timer Values (sec)
Reg Retry Intvl should be 120 seconds
https://support.freephoneline.ca/hc/en- ... redentials

save settings

7. Mango suggests the SIP T1 default setting is too aggressive, and to help resolve potential registration issues, T1 should be set to 1.

Navigate to Voice-->SIP-->SIP Timer Values-->SIP T1
Change SIP T1 to 1

save settings

8. Register Expires in your ATA must be set to 3600 seconds (page 6 of PDF guide)


If your ATA makes more than 5 registration attempts in 5 minutes (each time your reboot your ATA, it's attempting to register with Freephoneline; if the reg retry interval or Register Expires is too low, you also run the risk of getting temporarily IP banned), you may end up being temporarily IP banned by the specific FPL server the ATA was sending registration requests to. If you're temporarily IP banned, you could then try switching Proxy to a different FPL server than the one you were previously using
(voip.freephoneline.ca, voip2.freephoneline.ca, or voip4.freephoneline.ca:6060), unless you need to use voip4.freephoneline.ca:6060 because you have SIP ALG forced on in your router. The purpose of voip4.freephoneline.ca:6060 is to circumvent (buggy) SIP ALG features in routers.


From https://community.freepbx.org/t/trunk-s ... ca/22479/8
"As May 2013, our servers will rate limit REGISTER requests to a maximum of 10 requests per 5 minutes. Each authentication round usually consumes 2 requests (digest auth), so it is a fair number given our guidelines. Also, it does not affect INVITES (which are also authenticated)...

This rate limit is applied per IP address as our service is tailored to residential Canadian users (ADSL/Cable)."

If you're temporarily IP banned, you could then try switching Primary SIP Server to a different FPL server than the one you were previously using (voip.freephoneline.ca, voip2.freephoneline.ca, or voip4.freephoneline.ca:6060), unless you need to use voip4.freephoneline.ca:6060 because you have SIP ALG forced on in your router. The purpose of voip4.freephoneline.ca:6060 is to circumvent SIP ALG features in routers. Or you can disable FPL SIP registration in all devices (turn off your ATA) and wait a few hours until the temporary ban clears.

I have the PAP2T configured with voip.freephoneline.ca and my username and password but when I try and make a phone call, I get a busy signal everytime
When I call into my line, it goes straight to voicemail.

9. Always check the registration state both in your ATA (Info-->Line (FPL) status)and after logging in at https://www.freephoneline.ca/showSipSettings

Your ATA needs to be registered; SIP status needs to be connected; and if you see a SIP User Agent in your FPL account portal that you don't recognize, you've been hacked.

Note that only one registration per FPL account is allowed at any time. When there are multiple devices/softphones using the same account, only the most recent registration is valid. The previous device will lose registration. 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 with another device). Registration is required for incoming calls. It is not required for outgoing calls. If you simply want to make outgoing calls using your FPL number, configure, but don't register the account, on the SIP app being used. This is also important to consider if you're using Freephoneline's desktop application (don't have it running while using your ATA with the same FPL account). Additionally, keep in mind that if someone else is also attempting to register the same SIP credentials on another device where you live, too many registration attempts can result in a temporary IP ban.
Always check registration status in the ATA and also your SIP status after logging in at the above link.
If you see a device listed under SIP User Agent that you don't recognize, you've either been hacked or someone else is using your Freephoneline SIP username and SIP Password.

To help avoid being hacked don't port forward and don't use DMZ in your router. Do not use default device passwords for any devices connected to the the internet.




10. Login at https://www.freephoneline.ca/voicemailSettings, and ensure "Rings before voicemail" is greater than 1.


11. Log into your PAP2T

A. Under User 1 and User 2 tabs (both of these)

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)

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.

E. Ensure that you're using the correct SIP Username and Password: login at https://www.freephoneline.ca/showSipSettings.

I have also ensured I have UDP ports open 5060, 5061, 6060, 6061, 13000 and 13001.

12. Don't port forward unless you have no other choice, and especially don't use DMZ.

Other than 5060 and 6060 (6060 is only being used if you're using voip4.freephoneline.ca:6060), those aren't the right ports. Also, port forwarding is a security risk and should be avoided unless all else fails.
UDP ports 5061, 6061, 13000 and 13001 aren't used by your ATA in conjunction with FPL.

UDP ports 5060 (or 6060 depending on whether you're using voip.freephoneline.ca:6060 to try to circumvent buggy SIP ALG issues in routers) and RTP (UDP) port range 16384-16482 are. For the RTP ports, that range can be found in your ATA under SIP-->RTP Parameters-->RTP Port Min and RTP Port Max. The other UDP port being used is whatever local SIP port you defined in step 4. Again, port forwarding (especially UDP 5060) is a security risk and a great way to get hacked.


13. Proper device reboot order is always modem (wait for it to be fully up and running)–>router (wait for Wi-Fi SSIDs to populate first)–>ATA (in that order). Please reboot your devices in that order now.

[Liptonbrisk]

By the way, you appear to have upgraded your router at some point over the years.
You might want to take note of the following information, in particular, point #4 in the future.



Typically, for VoIP SIP services, especially for freephoneline, you want

1) a router that does not have a full cone NAT,

Visit https://www.think-like-a-computer.com/2 ... es-of-nat/.
Mango from the Obitalk.com forums writes,
“Use a restricted cone NAT router, and do not use port forwarding or DMZ. Restricted cone NAT will only permit
inbound traffic from the service provider you're registered to. If you have a full cone NAT router, it will allow traffic
from any source. This is probably not what you intend.
If you have a Windows computer, you can test your router using the utility here:
http://www.dslreports.com/forum/remark,22292023. To run it, use stun stun.ekiga.net from a command prompt.”
Essentially, you download the stun-test.zip file; extract the stun.exe file from within the zip file to an easily
accessible location; use an elevated command prompt (visit
http://www.thewindowsclub.com/how-to-ru ... inistrator); change directory (cd) to the
directory or location where you extracted stun.exe (visit
http://www.digitalcitizen.life/command- ... c-commands); and type “stun stun.ekiga.net” without
the quotation marks followed by the enter/return button on your keyboard.
Asus routers, at the time of this writing, produce port restricted cone NAT routers, for example and are fine,
provided you’re using one with Asuswrt-Merlin, third party firmware installed.

2) a router that lets you disable SIP ALG if it's buggy,

To understand why SIP ALG often causes horrible problems, please visit
http://www.voip-info.org/wiki/view/Routers+SIP+ALG (scroll down to the section on SIP ALG problems).

If you're dealing with a modem/router combo issued by an ISP or a router with SIP ALG forced on, you may have
to use voip4.freephoneline.ca:6060 for the Proxy Server. The purpose of voip4.freephoneline.ca:6060 is to circumvent
faulty SIP ALG features in routers.

3) a router that allows you to set QoS or assign highest priority to your ATA or IP Phone over all other devices on your LAN (local area network),

For a very general description of what QoS can do for you, visit https://www.voipmechanic.com/qos-for-voip.htm.
The basic idea is if you're torrenting or have a bunch of other computers, smartphones, tablets, etc. downloading and uploading (hogging all your available bandwidth), you don't want
your ATA not to have access to enough bandwidth to make or receive calls properly. So QoS or a Bandwidth Monitor feature (which is just another form of QoS) is a really good idea for VoIP users.

I often get an occasional relative complaining to me, "Hey my calls sound choppy." And then when I go visit, some kids are playing MMOs on a computer, while another person is downloading a huge file,
and another person is backing up files to a cloud service all at the same time someone else is trying to talk on the phone. All those devices, without QoS enabled, are fighting over available bandwidth along with the ATA.

and 4) A router that lets you adjust both Unreplied and Assured UDP timeouts.

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) < NAT Keep-alive Interval (in your ATA; for Obihai ATAs this is X_KeepAliveExpires) < UDP Assured Timeout (in your router) < SIP Registration Failure Retry Wait Time (or RegisterRetryInterval in Obihai ATAs)

“<“ 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, 10, 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, X_Keepalives expires is supposed to be 20 with FPL.

(the above settings are making reference to those in Obihai ATAs)

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 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 ... #p28056619.

The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 10 for UDP Unreplied Timeout and 117 for UDP Assured Timeout.



ISPs do not issue customers routers that can do all four things I just listed. Typically it's far better to have your own router with strong QoS functions and a restricted cone NAT firewall,
disable whatever SIP ALG feature is enabled in the router, and stick whatever modem/router combo your ISP gives you into bridge mode. For Bell Hubs, visit http://forums.redflagdeals.com/please-s ... r-1993629/. For Rogers, visit https://www.rogers.com/customer/support ... ridgemodem.