$100 VOIP unlock fee seems a little steep to use your own analog telephone adapter.
freephoneline.ca - Free Local Soft Phone Line for lifetime VOIP
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- SCORE+147
- PeZzy
- Jr. Member
- Mar 12, 2008
- 110 posts
- 42 upvotes
- Vancouver, BC
- embguy
- Deal Guru
- Sep 21, 2005
- 13456 posts
- 12205 upvotes
I paid $50 eight years ago. I spent total of $100 including the unlock key, porting my landline # and an used ATA box. It works out to $12.5 per year after eight year.
Everything are expensive nowadays. The earlier to decided to go with FPL, You will start saving money.
How much are you paying for a home phone service currently?
Do your calculation and make your decision.
If you continue to use what you were using, you will continue to pay what you were paying.
Have too many phones... This is how I limit my monthly phone payment.
Public Mobile $34 15GB, $11 250MB, 2x $5 50min/50text, $0 FPL home phone,
Fido $5 4GB plan with a free tablet.
Public Mobile $34 15GB, $11 250MB, 2x $5 50min/50text, $0 FPL home phone,
Fido $5 4GB plan with a free tablet.
- AsianXL
- Deal Addict
- Dec 18, 2007
- 4982 posts
- 6119 upvotes
You're probably paying like $40/month right now with a traditional phone line.
In about 6 months, you'll be pulling even.
I've been with FPL for over 10 years.
I have saved about $5000 already.
- cs993232
- Jr. Member
- Jul 29, 2013
- 104 posts
- 69 upvotes
- Toronto
a quick question and much appreciated for your help:
I have been facing the issue while the call-out is being perfect, the call-in only accepts my cellphone #, any other call ends up with voice mail. Digitmap is the regular one so I am puzzled.
I have been facing the issue while the call-out is being perfect, the call-in only accepts my cellphone #, any other call ends up with voice mail. Digitmap is the regular one so I am puzzled.
- ArrowFlynn
- Deal Addict
- Feb 6, 2020
- 1306 posts
- 1486 upvotes
Start here by answering the questions: https://forum.fongo.com/viewtopic.php?f=8&t=20199.
Digitmap has absolutely nothing to do with incoming calls.
If the incoming call problem is only with Rogers (Fido) carrier calls, the issue is SIP ALG (in most routers) or SIP Passthrough in Asus routers. Get it disabled.
- cs993232
- Jr. Member
- Jul 29, 2013
- 104 posts
- 69 upvotes
- Toronto
Works!, Thanks a million!ArrowFlynn wrote: ↑ Start here by answering the questions: https://forum.fongo.com/viewtopic.php?f=8&t=20199.
Digitmap has absolutely nothing to do with incoming calls.
If the incoming call problem is only with Rogers (Fido) carrier calls, the issue is SIP ALG (in most routers) or SIP Passthrough in Asus routers. Get it disabled.
- ArrowFlynn
- Deal Addict
- Feb 6, 2020
- 1306 posts
- 1486 upvotes
- Temporel
- Deal Guru
- Apr 10, 2011
- 12716 posts
- 25640 upvotes
- Montreal
I know this is a FPL thread but some are also using VoIP.ms
Update: Denver2 works fine for me.
denver2.voip.ms or 64.27.52.226
I haven't tested other POP.
Don't publicize it. Keep it for RFDers to prevent attacks.
You can also set a call forward from the site main page:
https://voip.ms/m/index.php
Update: Denver2 works fine for me.
denver2.voip.ms or 64.27.52.226
I haven't tested other POP.
Don't publicize it. Keep it for RFDers to prevent attacks.
You can also set a call forward from the site main page:
https://voip.ms/m/index.php
- ArrowFlynn
- Deal Addict
- Feb 6, 2020
- 1306 posts
- 1486 upvotes
There’s a dedicated thread about the VoIP.ms DDoS attack here: voip-ms-down-dns-ddos-attack-2489800/.
- Leshita
- Deal Addict
- Oct 15, 2003
- 2315 posts
- 578 upvotes
- Vancouver
I am currently on a $10 VOIP service with another provider.
I plan to finally dive in and get the freephoneline unlock key and pay to transfer my number.
There should be no difference between transferring a traditional analog line and transferring a VOIP line, right?
I just need to pay the $25 number transfer fee? I just want to double check before placing my order on freephoneline's Website.
I plan to finally dive in and get the freephoneline unlock key and pay to transfer my number.
There should be no difference between transferring a traditional analog line and transferring a VOIP line, right?
I just need to pay the $25 number transfer fee? I just want to double check before placing my order on freephoneline's Website.
- bylo
- Deal Expert
- Jan 7, 2002
- 29582 posts
- 28905 upvotes
- Waterloo, ON
Right. In theory. But I've seen reports that sometimes VoIP lines can be harder to port. I would contact the new provider, give them the phone number and get them to confirm that they can port it.
veni, vidi, Visa
- ArrowFlynn
- Deal Addict
- Feb 6, 2020
- 1306 posts
- 1486 upvotes
Visit https://support.freephoneline.ca/hc/en- ... r-porting- for more information.
If you want to use Freephoneline with an ATA, IP Phone, or SIP App, you also need to pay $99.95+tax for the VoIP unlock key. There's no free technical support.I just need to pay the $25 number transfer fee?
I suggest reading this guide even if you don't own an Obihai OBi2xx series ATA: https://forum.fongo.com/download/file.php?id=2164. If you're looking to buy a new ATA, take a look at http://forum.fongo.com/viewtopic.php?f= ... 805#p76546. In my opinion, Obihai ATAs are the most useful to use with FPL because they help to make up for FPL's lack of features compared to some other providers.
You might want to try the FPL desktop app on a PC before buying anything.
Here's how to install: http://forum.fongo.com/viewtopic.php?f= ... 810#p74810.
For the Freephoneline Windows desktop app . . .
Make sure that you're not muting anything (microphone/speakers), and that you tested to ensure your mic is working before fiddling around with the Windows app: http://win10faq.com/fix-microphone-settings/
And make sure you test incoming calls for 1-way audio issues before paying anything to FPL (you'll need a mic and headphones/speakers to test). Test on a computer that's connected to your router (without DMZ or port forwarding enabled). Should you encounter 1-way audio issues, look for a feature called SIP ALG in your router (you may need to call your ISP if you're using a modem/router combo) and disable that feature.
Steps i,ii, and iv below are for help dealing with 1-way audio issues with Freephoneline Windows desktop application.
from http://forums.redflagdeals.com/fongo-at ... #p27011164
Webslinger wrote: You can try the Freephoneline desktop app for free: https://www.freephoneline.ca/downloadDesktopApplication
It requires 32-bit Java to run. If you have problems installing the desktop app, visit http://forum.fongo.com/viewtopic.php?f= ... 63&p=74810.
A.Use winmtr https://sourceforge.net/projects/winmtr/
B. For Freephoneline.ca (based in Ontario), test to voip.freephoneline.ca (let winmtr ping about 100 times), voip2.freephoneline.ca, and voip4.freephoneline.ca. You can copy text to clipboard and paste your results (do not post your own IP public address though) and post them for others to examine if you want.
C. Look at the very last hop or line. Take a look at your average ping--and your maximum. You want those values to be relatively close.
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), you should probably avoid FPL.
I get between 11 (voip.freephoneline.ca and voip2.freephoneline.ca)-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.
Try the free FPL desktop app first: https://www.fongo.com/app/desktop/
Make sure that you're not muting anything (microphone/speakers), and that you tested to ensure your mic is working before fiddling around with the app: http://win10faq.com/fix-microphone-settings/
And make sure you test incoming calls for 1-way audio issues. Test on a computer that's connected to your router (without DMZ or port forwarding enabled). Should you encounter 1-way audio issues, look for a feature called SIP ALG in your router (you may need to call your ISP if you're using a modem/router combo) and disable that feature. You do need to do this in the ISP's device (better still to put it in bridge mode if you have your own router).
i. Typically it's better to have your own router and to stick whatever modem/router combo your ISP gives you into bridge mode.
ii. Disable SIP ALG in your own router. Many modem/router combos that are issued by ISPs have faulty SIP ALG/SPI functions enabled, with no way to disable them. These features can mangle SIP headers. If you don't know how to disable SIP ALG, contact your router's brand or contact your ISP.
To understand why SIP ALG is often a serious headache visit https://www.voip-info.org/routers-sip-alg/ (scroll down to "SIP ALG Problems")
iii. Properly enable QoS in your router for your computer that's running the Freephoneline desktop app (and ensure no other programs are running on your computer that are hogging bandwidth while using the Freephoneline desktop app). Refer to your router's manual or contact your ISP if you were issued a modem/router combo from them (typically those routers suck and have horrible or absent QoS functions).
I'm not a huge fan of this website, but it suffices for an explanation of QoS: http://www.voipmechanic.com/qos-for-voip.htm
Avoid anything it says about the G.729
iv. If you still get one-way audio issues with the Freephoneline desktop app, you may need to port forward, which is a security risk (and not advisable).
The FPL desktop app uses ports 5060-5061,6060-6061,13000-13001 if you're going to port forward for the desktop app (you need to port forward to the LAN IP of the computer you're using. For most home networks the IP will begin 192.168.xxx.x). Refer to your router's manual to learn how to port forward (if your router came from your ISP, contact your ISP).
I would start just by port forwarding 13000-13001 only, which is for RTP (audio packets). If that still doesn't work, you can try adding 6060 or 6061. The most dangerous ports to forward are 5060-5061 and really shouldn't be necessary if you're forwarding 6060 or 6061 anyway. I guess if all else fails, forward all of them: 5060, 5061,6060, 6061,13000, and 13001.
These are all UDP ports.
5060, 5061, 6060, and 6061 should be alternate SIP ports.
Only port forward if all else fails (and only do it temporarily, since it's a security risk).
Last edited by ArrowFlynn on Sep 25th, 2021 3:58 pm, edited 1 time in total.
- ArrowFlynn
- Deal Addict
- Feb 6, 2020
- 1306 posts
- 1486 upvotes
http://forum.fongo.com/viewtopic.php?f=8&t=20199#p78976
(taken from Webslinger originally)
(Generic info)
Typically, for VoIP SIP services, especially for Freephoneline/Fongo, you want
A) a router that does not have a full cone NAT,
Visit https://dh2i.com/kbs/kbs-2961448-unders ... -punching/.
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.
B) a router that lets you disable SIP ALG if it's buggy,
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).
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.
C) 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 D) 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; for Grandstream, the setting is SIP OPTIONS Keep Alive Interval) < 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, 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, X_KeepAlivesExpires (SIP OPTIONS Keep Alive Interval) 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 (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: recommendations-new-router-2115672/2/#p28059363.
Router firmware that allows users to adjust Assured and Unreplied UDP timeouts include
Asuswrt-Merlin
Ubiquiti
Mikrotik
pfSense
Tomato
DD-WRT
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.
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.
- Leshita
- Deal Addict
- Oct 15, 2003
- 2315 posts
- 578 upvotes
- Vancouver
Thanks, I will try to figure out a way to contact freephoneline.
ArrowFlynn wrote: ↑ Visit https://support.freephoneline.ca/hc/en- ... r-porting- for more information.
If you want to use Freephoneline with an ATA, IP Phone, or SIP App, you also need to pay $99.95+tax for the VoIP unlock key. There's no free technical support.
I suggest reading this guide even if you don't own an Obihai OBi2xx series ATA: https://forum.fongo.com/download/file.php?id=2164. If you're looking to buy a new ATA, take a look at http://forum.fongo.com/viewtopic.php?f= ... 805#p76546. In my opinion, Obihai ATAs are the most useful to use with FPL because they help to make up for the FPL's lack of features compared to some other providers.
You might want to try the FPL desktop app on a PC before buying anything.
Here's how to install: http://forum.fongo.com/viewtopic.php?f= ... 810#p74810.
For the Freephoneline Windows desktop app . . .
Make sure that you're not muting anything (microphone/speakers), and that you tested to ensure your mic is working before fiddling around with the Windows app: http://win10faq.com/fix-microphone-settings/
And make sure you test incoming calls for 1-way audio issues before paying anything to FPL (you'll need a mic and headphones/speakers to test). Test on a computer that's connected to your router (without DMZ or port forwarding enabled). Should you encounter 1-way audio issues, look for a feature called SIP ALG in your router (you may need to call your ISP if you're using a modem/router combo) and disable that feature.
Steps i,ii, and iv below are for help dealing with 1-way audio issues with Freephoneline Windows desktop application.
from http://forums.redflagdeals.com/fongo-at ... #p27011164
ArrowFlynn wrote: ↑ http://forum.fongo.com/viewtopic.php?f=8&t=20199#p78976
(taken from Webslinger originally)
(Generic info)
Typically, for VoIP SIP services, especially for Freephoneline/Fongo, you want
A) a router that does not have a full cone NAT,
Visit https://dh2i.com/kbs/kbs-2961448-unders ... -punching/.
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.
B) a router that lets you disable SIP ALG if it's buggy,
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).
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.
C) 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 D) 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; for Grandstream, the setting is SIP OPTIONS Keep Alive Interval) < 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, 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, X_KeepAlivesExpires (SIP OPTIONS Keep Alive Interval) 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 (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: recommendations-new-router-2115672/2/#p28059363.
Router firmware that allows users to adjust Assured and Unreplied UDP timeouts include
Asuswrt-Merlin
Ubiquiti
Mikrotik
pfSense
Tomato
DD-WRT
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.
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.
Wow, thank guys for directing me to these guides.
At the moment I have a Grandstream HT812 ATA and a TP-Link AC1350.
Looks like I am going to have to go through these to see whether I will need new gear.
- ArrowFlynn
- Deal Addict
- Feb 6, 2020
- 1306 posts
- 1486 upvotes
Your ATA will work, but for users who want to work around some of FPL’s limitations, an Obihai ATA would be a better choice.
It’s unnecessary to buy new stuff unless you encounter problems first.
Last edited by ArrowFlynn on Sep 25th, 2021 3:35 pm, edited 1 time in total.
- Leshita
- Deal Addict
- Oct 15, 2003
- 2315 posts
- 578 upvotes
- Vancouver
I bought that ATA last time when it was on sale at BB without researching first.ArrowFlynn wrote: ↑ Your ATA will work, but for users who want to make the most of FPL’s limitations, an Obihai ATA would be a better choice.
Not ideal since you can’t adjust UDP timeouts, but it’s unnecessary to run around buying new stuff unless you encounter problems first.
I guess I will try it out with my current gears first.
I have noticed that there are occasional monthly disconnects with my current provider, hopefully it is my provider and not the gear itself.
Thanks for the tips!
- ArrowFlynn
- Deal Addict
- Feb 6, 2020
- 1306 posts
- 1486 upvotes
That could be due to any number of things, including your ISP.
If you notice the problem happens when your public WAN IP changes (modem IP lease renews), the problem is likely associated with UDP timeouts in your router, which you can’t adjust using your official router firmware.
NAT corruption can occur in routers without user intervention.
If your internet service is unreliable or if you're dealing with corrupted NAT associations that eventually develop between your router and ATA, switching to other SIP service providers will not help you.
Look at pages 44 to 46 in the guide I linked earlier about losing registration.
FPL is, in my opinion, not a good idea for beginners. You will be your own tech support.
Keep in mind that FPL’s servers are located in Southern Ontario. The closer you are, generally speaking, the better the experience should be with respect to latency (lag) and jitter (choppiness and disconnects).
Last edited by ArrowFlynn on Sep 25th, 2021 6:16 pm, edited 2 times in total.
- bylo
- Deal Expert
- Jan 7, 2002
- 29582 posts
- 28905 upvotes
- Waterloo, ON
You may want to try something like TekSavvy's TekTalk ($10 to $20 per month.) They use the HT812A. So even if you bought the ATA at BB they should be able to supply you with a working config file that you can import into the ATA.Leshita wrote: ↑ I bought that ATA last time when it was on sale at BB without researching first.
I guess I will try it out with my current gears first.
I have noticed that there are occasional monthly disconnects with my current provider, hopefully it is my provider and not the gear itself.
Thanks for the tips!
The advantage of TekSavvy is they provide good technical support. That will get you up and running. It will also get you familiar with VoIP in general. Then decide if you're happy paying TekSavvy's prices or if you want to switch to a cheaper alternative like FPL or voip.ms.
veni, vidi, Visa
- Leshita
- Deal Addict
- Oct 15, 2003
- 2315 posts
- 578 upvotes
- Vancouver
ArrowFlynn wrote: ↑ That could be due to any number of things, including your ISP.
If you notice the problem happens when your public WAN IP changes (modem IP lease renews), the problem is likely associated with UDP timeouts in your router, which you can’t adjust using your official router firmware.
NAT corruption can occur in routers without user intervention.
If your internet service is unreliable or if you're dealing with corrupted NAT associations that eventually develop between your router and ATA, switching to other SIP service providers will not help you.
Look at pages 44 to 46 in the guide I linked earlier about losing registration.
FPL is, in my opinion, not a good idea for beginners. You will be your own tech support.
Keep in mind that FPL’s servers are located in Southern Ontario. The closer you are, generally speaking, the better the experience should be with respect to latency (lag) and jitter (choppiness and disconnects).
This is quite a bit of info to take in. I am currently paying $10 per month with AEBC. I am not sure when it disconnects but I have noticed it does do it quite often, usually I have to unplug the ATA and replug before it resumes service.bylo wrote: ↑ You may want to try something like TekSavvy's TekTalk ($10 to $20 per month.) They use the HT812A. So even if you bought the ATA at BB they should be able to supply you with a working config file that you can import into the ATA.
The advantage of TekSavvy is they provide good technical support. That will get you up and running. It will also get you familiar with VoIP in general. Then decide if you're happy paying TekSavvy's prices or if you want to switch to a cheaper alternative like FPL or voip.ms.
The next time I will try and see if my public WAN IP changes. Thank you guys.
- ArrowFlynn
- Deal Addict
- Feb 6, 2020
- 1306 posts
- 1486 upvotes
Daily? Hourly? Is the interval exactly the same each time?
In the end, I’ve never used AEBC, but it sounds as though you had a locked ATA before. I have no clue what settings they’re using for NAT Keep Alive or registration timers.
You could check to see whether the problem goes away when connecting directly to your modem instead of the TPLink router, but if the modem doesn’t also contain a a NAT firewall router, the ATA will be unprotected. If the problem doesn’t continue while connected to the modem, the problem involves your TPLink router.
NAT firewall (UDP port closing after a certain period) or UDP timeout issues can cause the problem you’re describing. Possibly port forwarding may help (you would have to contact AEBC to find out what ports are being used), but port forwarding is a security risk, particularly if you have to port forward UDP port 5060 to address this issue.