Freebies

freephoneline.ca - Free Local Soft Phone Line for lifetime VOIP

Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3260 upvotes
softech wrote:
Dec 26th, 2018 2:44 pm
does freephoneonkinr allows multiple logon at the same time??
No, and that matters for incoming calls and temporary IP bans.

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. If you ever see a SIP user agent that you don't recognize after logging in at the above link, someone else is using your credentials (possibly, you've been hacked in that scenario).

Also, more than 5 registration attempts within 5 minutes can result in a temporary IP ban.

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). "
Last edited by Guest1284983 on Dec 28th, 2018 12:37 am, edited 1 time in total.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
Deal Addict
Mar 23, 2003
1258 posts
99 upvotes
Webslinger wrote:
Dec 27th, 2018 3:43 pm
No, but that only matters for incoming calls and temporary IP bans.

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.
what exactly mean configure but don't register the account?

my Samsung has the following SIP account details? which one to fill in so it wont register the account?

username: 1647xxxxxxx <- do I fill in this?
password: xxxxx <- do I fill in this?
server: voip.freephoneline.ca
Authentication username <- do I fill in this? or fill ein 1647xxxxxxx ?
outbound proxy address: voipfreephoneline.ca
portnumber 5060
Transport Type UDP

am I right?
thanks again
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3260 upvotes
softech wrote:
Dec 27th, 2018 11:35 pm
what exactly mean configure
You can enter SIP credentials in the app--but don't attempt FPL account registration.

In decent SIP apps, such as Acrobits Groundwire, you can for the account(s) (edit account menu) in question, select "Off -- Do Not Register" for incoming calls.

There's also a global setting (found under settings-->Incoming calls), which will affect your accounts if they're set to use to the global setting for incoming calls. So you may need to change that to “Off — Do Not Register” as well.

Additionally, "Outgoing calls need registration" needs to be disabled in Groundwire (Edit Account—>Advanced Settings). This setting should always be disabled for FPL even when not troubleshooting.

I can then make then make an outbound call using Groundwire on my smartphone using the same FPL account in my ATA that's currently registered. But during that call period, inbound calls to the ATA will not work. When I end the call in Groundwire, inbound calls to the ATA work again.

For the most part though, I'm calling into the Obihai ATA using its Auto Attendant to place outbound calls (typically for long distance calling only) because jitter and pings usually increase over cellular data.


but don't register the account?
https://andrewjprokop.wordpress.com/201 ... istration/


You can't have your FPL account simultaneously registered in a SIP app (your smartphone) and in your ATA. Only the most recent registration will be valid. The older registration will not.
You may only register your FPL account on one device at any given time. Only one registration per FPL account is allowed at any time. The most recently registered device will ring for incoming calls. The previously registered device will not. Registration is required for incoming calls. Registration is not required for outgoing calls.

Everytime your ATA reboots (or approximately every 3600 seconds, if it's configured properly, at the next registration interval), your ATA attempts to register with FPL.
You phone's app will also likely attempt to register with FPL, unless you specify otherwise (I am not familiar with all SIP apps; possibly some don't offer the option to not register).
If you attempt more than 5 registrations within 5 minutes, you may be temporarily IP banned from the FPL Proxy Server you were attempting to register with. And then that Proxy Server will block you from attempting to do anything on it until the ban clears.

https://support.freephoneline.ca/hc/en- ... redentials
If you're unable to specify a registration period (or approximate interval) of 3600 seconds or 1 hr, you're going to run the risk of being temporarily IP banned, should you choose to register your FPL SIP credential at shorter intervals.

With respect to temporary IP bans, I did make a related change in the PDF guide on page 22: http://forum.fongo.com/viewtopic.php?f= ... 805#p73839.


my Samsung has the following SIP account details? which one to fill in so it wont register the account?

username: 1647xxxxxxx <- do I fill in this?
password: xxxxx <- do I fill in this?
server: voip.freephoneline.ca
Authentication username <- do I fill in this? or fill ein 1647xxxxxxx ?
outbound proxy address: voipfreephoneline.ca
portnumber 5060
Transport Type UDP
You shouldn't need to specify an outbound proxy address since it's the same as the main proxy server (and even if you did, you would need voip.freephoneline.ca and not voipfreephoneline.ca), but it's possible the app forces you to enter it. Yes, you do need to specify password. Authentication username should definitely be what you find after logging in at https://www.freephoneline.ca/showSipSettings. Username should also be the same phone number.

I would disable anything related to outbound proxy. And I would choose not to register the account if that option is available. If you have no choice but to register, then you're going to run into problems if your ATA is registered with FPL simultaneously. Nothing you've specified indicates the app won't register with FPL's proxy server.

Using Fongo Mobile, which is exactly the same as FPL, with the exception of the phone number provided by FPL, probably makes more sense, provided the Fongo Mobile app works reasonably well for you. Fongo Mobile and FPL accounts are separate. It doesn't matter if you use Fongo Mobile simultaneously with FPL.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
Newbie
Dec 23, 2012
73 posts
76 upvotes
Hey Webslinger,

I am new the VOIP game - I have an ASUS AC1900/AC68U setup with Bell DSL, how can I test if freephoneline with an ATA (such as Obihai 202/200) will work before buying their $90 login/password?

Looking to replace home phone with a cheaper alternative, no need for US-calls; Canada-wide is good enough.

Could you send me some good reading material as well?

Thank you in advance.
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3260 upvotes
Trance wrote:
Dec 30th, 2018 12:27 am
Hey Webslinger,

I am new the VOIP game - I have an ASUS AC1900/AC68U setup with Bell DSL
Using Asuswrt-Merlin, third party firmware for Asus routers, would be a good idea (I accept no responsibility for issues that may arise from using third party firmware, such as warranty issues or bricked hardware due to user error from failed firmware updates) : https://asuswrt.lostrealm.ca/about. I also suggest setting up Bandwidth Monitor (assigning highest priority to an Obihai ATA once you get one), and changing UDP timeouts (once you get an Obihai ATA), and doing PPPoE login using Asuswrt-Merlin (your router).


how can I test if freephoneline with an ATA (such as Obihai 202/200) will work before buying their $90 login/password?

You can try the desktop app for free: http://forums.redflagdeals.com/rogers-h ... #p29235388.
If you encounter problems installing it, visit http://forum.fongo.com/viewtopic.php?f= ... 810#p74810.
Could you send me some good reading material as well?

The first two pages of this thread would probably be a good starting point: newegg-obihai-obi200-ata-49-99-1-50-ehf ... x-2145415/.
I would also encourage you to read this PDF guide: http://forum.fongo.com/viewtopic.php?f= ... 805#p73839.


---
For anyone new to using SIP services as home phones, I'm just going to copy and paste something I've written before (mostly within the context of helping FPL users, but all four points apply to users of home phone SIP services in general). There's too many newcomers (and even existing users) who don't know this information.




For VoIP SIP services, 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: https://asuswrt.lostrealm.ca/about.

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, 17, 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 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:
recommendations-new-router-2115672/2/#p28056619 (I've never used them and can't advise buying them or answering questions about them)
recommendations-new-router-2115672/2/#p28059363
recommendations-new-router-2115672/2/#p28059444

The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 17 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 you can just do PPPoE login using your own router if you want). For Bell Hubs, visit http://forums.redflagdeals.com/please-s ... r-1993629/. For Rogers Hitron, visit https://www.rogers.com/customer/support ... ridgemodem (CGN3 instructions also apply to CODA-4582).
Last edited by Guest1284983 on Jan 9th, 2019 2:41 pm, edited 2 times in total.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
Deal Addict
Jun 29, 2010
1128 posts
508 upvotes
Somewhere in Ontario
How is the 911 fee charged? Do I get a bill in the mail or email? How do I pay it? My mom slipped and fell and got a good goose egg so she needed to get checked out.
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3260 upvotes
DPyro wrote:
Dec 30th, 2018 4:17 pm
How is the 911 fee charged? Do I get a bill in the mail or email? How do I pay it? My mom slipped and fell and got a good goose egg so she needed to get checked out.
Sorry about your mom. I hope she's okay!

Were you actually charged though? My impression is that the $35 fee is just a threat to make sure people don't test 911 calls. The threat scares me enough that I never make test 911 calls anymore.
After you log in, do you see a fee for the 911 call under the "cost" column? Login at https://www.freephoneline.ca/doGetCallLogs.

Look at what @__wizard__ wrote over here: https://forums.redflagdeals.com/freepho ... #p27964332.

Anyway, if you were invoiced or charged, submit a Billing and Payment inquiry: https://support.fongo.com/hc/en-us/requests/new. They don't work holidays or weekends.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
Deal Addict
Jun 29, 2010
1128 posts
508 upvotes
Somewhere in Ontario
Cost shows $0.00
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3260 upvotes
DPyro wrote:
Dec 30th, 2018 5:25 pm
Cost shows $0.00
Thanks for the info! I wouldn’t sweat it too much unless you’re invoiced or contacted about it. Just don’t make test 911 calls.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
Deal Fanatic
Jun 7, 2005
7879 posts
490 upvotes
I just came back from a month vacation, and noticed the FPL is not working. I can't make or receive calls. Incoming calls go to voicemail directly.

I just checked my FPL account. SIP Status shows "Disconnected"

Please help.

Thanks
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3260 upvotes
rdx wrote:
Jan 2nd, 2019 8:01 pm
I just came back from a month vacation, and noticed the FPL is not working. I can't make or receive calls. Incoming calls go to voicemail directly.

I just checked my FPL account. SIP Status shows "Disconnected"

Please help.

Thanks
You should also be checking registration status in the ATA.
a)Dial ***1. Enter the IP address you hear into a web browser.
b)Log into the ATA (default username and password are "admin" without quotation marks).
c)Navigate to Status-->System Status-->SP(FPL) Service Status
What does the registration status indicate?

Refer to pages 47 to 48 (do steps A to D) of the PDF guide: http://forum.fongo.com/viewtopic.php?f= ... 805#p73839

I would also advise going through the PDF guide again, since it’s been revised (especially page 22).

Again, there’s not much else I can do for you: https://forums.redflagdeals.com/freepho ... #p30154880.




The real fix is to get a router that meets the criteria below, especially #4. Otherwise, you're eventually going to need to remember the proper device reboot sequence whenever you have this problem:
modem-->router (wait for it to be fully up and running with Wi-Fi SSIDs populated first)--->ATA




For VoIP SIP services, 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: https://asuswrt.lostrealm.ca/about.

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, 17, 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 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 ... #p28056619 (I've never used them and can't advise buying them or answering questions about them)
https://forums.redflagdeals.com/recomme ... #p28059363
https://forums.redflagdeals.com/recomme ... #p28059444

The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 17 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 you can just do PPPoE login using your own router if you want). For Bell Hubs, visit please-sticky-how-bypass-bell-hub-use-y ... r-1993629/. For Rogers Hitron, visit https://www.rogers.com/customer/support ... ridgemodem (CGN3 instructions also apply to CODA-4582).
Last edited by Guest1284983 on Jan 2nd, 2019 9:39 pm, edited 1 time in total.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
Deal Fanatic
Jun 7, 2005
7879 posts
490 upvotes
FPL status says "Register Failed: No Response From Server (server=162.213.111.21:6060; retry in 6s)"
Webslinger wrote:
Jan 2nd, 2019 9:16 pm
You should also be checking registration status in the ATA.
a)Dial ***1. Enter the IP address you hear into a web browser.
b)Log into the ATA (default username and password are "admin" without quotation marks).
c)Navigate to Status-->System Status-->SP(FPL) Service Status

What does the registration status indicate?

Refer to pages 47 to 48 (do steps A to D) of the PDF guide: http://forum.fongo.com/viewtopic.php?f= ... 805#p73839

I would also advise going through the guide again, since it’s been revised (especially page 22).

Again, there’s not much else I can do for you: https://forums.redflagdeals.com/freepho ... #p30154880




The real fix is to get a router that meets this criteria, especially #4. Otherwise, you're eventually going to need to remember the proper device reboot sequence:
modem-->router (wait for it to be fully up and running with Wi-Fi SSIDs populated first)--->ATA




For VoIP SIP services, 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: https://asuswrt.lostrealm.ca/about.

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, 17, 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 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 ... #p28056619 (I've never used them and can't advise buying them or answering questions about them)
https://forums.redflagdeals.com/recomme ... #p28059363
https://forums.redflagdeals.com/recomme ... #p28059444

The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 17 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 you can just do PPPoE login using your own router if you want). For Bell Hubs, visit please-sticky-how-bypass-bell-hub-use-y ... r-1993629/. For Rogers Hitron, visit https://www.rogers.com/customer/support ... ridgemodem (CGN3 instructions also apply to CODA-4582).
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3260 upvotes
rdx wrote:
Jan 2nd, 2019 9:39 pm
FPL status says "Register Failed: No Response From Server (server=162.213.111.21:6060; retry in 6s)"
The lack of a response may suggest you've been temporarily IP banned by voip4.freephoneline.ca:6060. But I would go through steps A to D on pages
47 to 48 of the PDF guide regardless.


Refer to pages 47 to 48 (do steps A to D) of the PDF guide: http://forum.fongo.com/viewtopic.php?f= ... 805#p73839 .
Refer to the section called "Are Freephoneline’s SIP servers down? My ATA isn’t registered."

i) In your Obihai ATA or at Obitalk.com (whichever you normally use; don't use both), Navigate to Voice Services-->SP(FPL)
Service-->X_UserAgentPort. X_UserAgentPort should be a random port number between 30000 and
60000. Just pick a port number in that range. Change to a new port number in that range. Click the
“submit” button, and reboot the ATA. (If you use Obitalk.com to change settings, you will need to use
Obitalk.com).
If changing X_UserAgentPort works, you were dealing with a corrupted NAT connection in your router.

ii) Double check your Registration timers (refer to page 21). For RegistrationPeriod use 3600, and
RegisterRetryInterval should be 120. Set X_CheckPrimaryFallbackInterval to 7200 seconds.
If your ATA makes more than 5 registration attempts in 5 minutes,
you may end up being temporarily IP banned by the specific FPL server the ATA was sending
registration requests to. Each time you reboot your ATA, it's making a registration attempt.
If you're temporarily IP banned, you could then try switching ProxyServer (refer
to pages 14, 20, and 21) 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.

Temporary IP bans may take a couple of hours to clear. During this period, ensure your ATA isn't attempting to register
with the exact same FPL proxy server over and over again. If you have no other choice but to use voip4.freephoneline.ca:6060 due to being
unable to disable SIP ALG in your router, then keep your ATA unplugged for a couple of hours. After a couple of hours, refer to the proper device reboot sequence:
modem-->router (wait for it to be fully up and running with Wi-Fi SSIDs populated first)--->ATA.



I would also advise going through the PDF guide again, since it’s been revised (especially page 22 for X_CheckPrimaryFallbackInterval).


Again, there’s not much else I can do for you: https://forums.redflagdeals.com/freepho ... #p30154880.




It's a good idea to get a router that meets the criteria below, especially #4. Otherwise, you're eventually going to need to remember the proper device reboot sequence:
modem-->router (wait for it to be fully up and running with Wi-Fi SSIDs populated first)--->ATA




For VoIP SIP services, 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: https://asuswrt.lostrealm.ca/about.

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, 17, 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 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 ... #p28056619 (I've never used them and can't advise buying them or answering questions about them)
https://forums.redflagdeals.com/recomme ... #p28059363
https://forums.redflagdeals.com/recomme ... #p28059444

The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 17 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 you can just do PPPoE login using your own router if you want). For Bell Hubs, visit please-sticky-how-bypass-bell-hub-use-y ... r-1993629/. For Rogers Hitron, visit https://www.rogers.com/customer/support ... ridgemodem (CGN3 instructions also apply to CODA-4582).
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
Deal Fanatic
Jun 7, 2005
7879 posts
490 upvotes
I reboot everything and it is working again. But I understand you explained that it is not the permanent solution.

Should I just simply switch to the modem/built in router (Hitron cgn3)provided by the internet service provider ? And it will solve the issue ?

Thanks

Webslinger wrote:
Jan 2nd, 2019 9:41 pm
The lack of a response may suggest you've been temporarily IP banned by voip4.freephoneline.ca:6060. But I would go through steps A to D on pages
47 to 48 of the PDF guide regardless.


Refer to pages 47 to 48 (do steps A to D) of the PDF guide: http://forum.fongo.com/viewtopic.php?f= ... 805#p73839 .
Refer to the section called "Are Freephoneline’s SIP servers down? My ATA isn’t registered."

i) In your Obihai ATA or at Obitalk.com (whichever you normally use; don't use both), Navigate to Voice Services-->SP(FPL)
Service-->X_UserAgentPort. X_UserAgentPort should be a random port number between 30000 and
60000. Just pick a port number in that range. Change to a new port number in that range. Click the
“submit” button, and reboot the ATA. (If you use Obitalk.com to change settings, you will need to use
Obitalk.com).
If changing X_UserAgentPort works, you were dealing with a corrupted NAT connection in your router.

ii) Double check your Registration timers (refer to page 21). For RegistrationPeriod use 3600, and
RegisterRetryInterval should be 120. Set X_CheckPrimaryFallbackInterval to 7200 seconds.
If your ATA makes more than 5 registration attempts in 5 minutes,
you may end up being temporarily IP banned by the specific FPL server the ATA was sending
registration requests to. Each time you reboot your ATA, it's making a registration attempt.
If you're temporarily IP banned, you could then try switching ProxyServer (refer
to pages 14, 20, and 21) 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.

Temporary IP bans may take a couple of hours to clear. During this period, ensure your ATA isn't attempting to register
with the exact same FPL proxy server over and over again. If you have no other choice but to use voip4.freephoneline.ca:6060 due to being
unable to disable SIP ALG in your router, then keep your ATA unplugged for a couple of hours. After a couple of hours, refer to the proper device reboot sequence:
modem-->router (wait for it to be fully up and running with Wi-Fi SSIDs populated first)--->ATA.



I would also advise going through the PDF guide again, since it’s been revised (especially page 22 for X_CheckPrimaryFallbackInterval).


Again, there’s not much else I can do for you: freephoneline-ca-free-local-soft-phone- ... #p30154880.




It's a good idea to get a router that meets the criteria below, especially #4. Otherwise, you're eventually going to need to remember the proper device reboot sequence:
modem-->router (wait for it to be fully up and running with Wi-Fi SSIDs populated first)--->ATA




For VoIP SIP services, 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: https://asuswrt.lostrealm.ca/about.

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, 17, 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 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:
recommendations-new-router-2115672/2/#p28056619 (I've never used them and can't advise buying them or answering questions about them)
recommendations-new-router-2115672/2/#p28059363
recommendations-new-router-2115672/2/#p28059444

The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 17 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 you can just do PPPoE login using your own router if you want). For Bell Hubs, visit http://forums.redflagdeals.com/please-s ... r-1993629/. For Rogers Hitron, visit https://www.rogers.com/customer/support ... ridgemodem (CGN3 instructions also apply to CODA-4582).

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