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

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Mar 3, 2002
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redkulat wrote:
May 16th, 2017 3:36 pm
This is very interesting, I wonder how much people are charging to make it a business. I actually thought about it too but figured it was way too much hassle to go through.
I don't know why most people just wouldn't read the guide I wrote and setup the service themselves.

I'd have to spend my time doing TeamViewer, Remote Desktop, or something similar (or worse, having to do in-home service). I have zero interest.
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OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
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Jan 27, 2004
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Webslinger,

I know freephoneline is under fongo but what is the diff btw them? How does freephoneine make $$?

These r the Q my friends ask
2007 - Ipod Video (TD), Ipod Shuffle (GM)
2006 - Ipod Nano (TD)
2005 - Ipod Shuffle (TD)
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Nov 27, 2004
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Gatineau, QC
I want to thank Webslinger for his terrific Obi202 set-up guide for FreePhoneLine.ca .

I have my FreePhoneLine account via the one time fee. It is up and running and today my POTS number was ported over. Everything works great and my wife is happy with the call quality (which means I am happy!).

Thanks for helping me save money.
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Cantley wrote:
May 26th, 2017 12:44 pm
I want to thank Webslinger for his terrific Obi202 set-up guide for FreePhoneLine.ca .

I have my FreePhoneLine account via the one time fee. It is up and running and today my POTS number was ported over. Everything works great and my wife is happy with the call quality (which means I am happy!).

Thanks for helping me save money.

That's awesome!!!
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.
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Cantley wrote:
May 26th, 2017 12:44 pm
I want to thank Webslinger for his terrific Obi202 set-up guide for FreePhoneLine.ca .
By the way, the latest version is now 1.36: http://forum.fongo.com/viewtopic.php?f= ... 624#p74624
I updated the questions on the final two pages.
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.
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Apr 28, 2010
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I noticed complete silence when dialing my number when it's not registered on any device using a sip device. Is there any way have it ring instead of silence when I disconnect from a device? Having silence will just make people hang up instead of waiting to leave a message.
"signature removed for rule violations" <--- Hell ya, cause I'm a rule breaker! 😝
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pete_rfd wrote:
May 27th, 2017 5:27 pm
I noticed complete silence when dialing my number when it's not registered on any device using a sip device. Is there any way have it ring instead of silence when I disconnect from a device? Having silence will just make people hang up instead of waiting to leave a message.
First if you have no registration, incoming calls should be going directly to FPL's voicemail (no ringing, no silence).

So, I can't reproduce what's happening to you.

If SIP status shows "disconnected" when I log in at https://www.freephoneline.ca/showSipSettings, incoming calls will go straight to voicemail.

If I disable the ATA, by unplugging it, incoming calls go straight to voicemail (there may be a 3 to 5s delay at first).

There's no lengthy silence or ringing.

I'm trying to think of a situation that would cause what you're experiencing to happen. Basically, FPL's server would need to think your ATA is still online and registered.
The call would then been directed to your ATA by FPL. The silence is ringing that's supposed to be taking place. After the required duration (as defined by "rings before voicemail") is met, the incoming call is redirected to voicemail.


Although there is a crazy huge registration period of 1 hour where the ATA does not communicate with FPL's server, Keep Alive packets are supposed to be transmitted.
The only thing that comes to mind off the top of my head is that your ATA may not really be offline. Possibly there was a brief internet outage and your ATA came back online with a corrupted NAT connection between your router and your ATA. I've discussed corrupted NAT connections caused by UDP timeout settings on page 3 (point #6) of this PDF guide located at http://forum.fongo.com/viewtopic.php?f= ... 805#p74624 and also over at http://forums.redflagdeals.com/need-rou ... #p27768592.

FWIW, if you log in at https://www.freephoneline.ca/voicemailSettings and have Rings Before Voicemail set to 1, all incoming calls, no matter what, should be going to voicemail immediately after the duration of 1 ring.

Probably the best person to answer your question is @PianoGuy, but he doesn't seem to post in this thread very often anymore.
Last edited by Guest1284983 on May 27th, 2017 6:36 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.
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Apr 28, 2010
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Wow @Webslinger, so much info there thank you for the reply.

Using the home phone, if I unplug the ATA from the power source, FPL will still say I'm registered on that device, and there won't be any rings. I was curious if the same would happen if I just unplug the phone cable from the ATA, still said registered to that device, but this time I heard rings before going to voicemail. Unplugging either the power or the phone cables, neither time does it go straight to voicemail like you said it should though, which is what I assume should be happening.

The only time it goes straight to voicemail is if I unplug the device overnight, and in the morning I tested a call, and it went straight to voicemail, but that's because registration had expired.
"signature removed for rule violations" <--- Hell ya, cause I'm a rule breaker! 😝
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Mar 3, 2002
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pete_rfd wrote:
May 27th, 2017 9:09 pm

Using the home phone, if I unplug the ATA from the power source, FPL will still say I'm registered on that device, and there won't be any rings. I was curious if the same would happen if I just unplug the phone cable
Do you mean Ethernet cable?

I noticed complete silence when dialing my number when it's not registered
Okay, you said FPL isn't registered in your initial post. Are you checking the registration status in the ATA to determine registration status?
If the ATA indicates FPL is registered when you're encountering this issue, you're likely dealing with a corrupted NAT connection that's developed between your router and the ATA (and in this scenario, it's possible that the caller may hear ringing on his or her end).
If the ATA indicates you're not registered but FPL's website indicates that you are, it's possible FPL's server still tries to send data to your WAN IP, but upon failure (not being able to connect) the call should be routed to voicemail. There may be a slight delay (maybe 3-5 seconds) before that happens, but I don't hear any ringing. I wouldn't consider 3-5 seconds (after clicking "dial") before being transferred to voicemail a huge deal given that it usually takes at least 10s of ringing before calls go to voicemail normally with regular calls anyway. I figure the average person will probably still be on the phone 5s after hitting "dial" even if there's silence.

Registration status via FPL's website usually lags behind what's actually going on in the ATA (registration interval is 3600 seconds, and registration status may not update in FPL's web portal until the current registration period expires). My only point is if the FPL web portal indicates you're disconnected, there's no way FPL's server is trying to send anything to your ATA. The call should certainly be sent directly to voicemail without any ringing.

Regardless, when my ATA is unplugged or unregistered, I don't hear rings. The incoming call goes to voicemail.
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.
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Jul 12, 2006
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I need ton port out my number in a hurry. What is my best option to keep the number for not much expense? Currently with koodo and don't really want to pay vacation hold while I am away.

Thanks.
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Ferris Bueller wrote:
May 28th, 2017 2:35 pm
I need ton port out my number in a hurry. What is my best option to keep the number for not much expense? Currently with koodo and don't really want to pay vacation hold while I am away.



I'm not sure your question has anything to do with Freephoneline. The RFD cell phone forum is located at cell-phones-f88/.

The cheapest option, in the long run, is to port to Fongo Mobile for $25+tax. There's no ongoing fees, and you can still use your mobile number on a smartphone using cellular data or wi-fi provided your phone number is in one of these cities: https://support.fongo.com/hc/en-us/articles/212436086. Fongo Mobile also supports SMS (FPL does not).

Click https://support.fongo.com/hc/en-us/arti ... e-accounts for more info.

The RFD Fongo Mobile thread is over here: merged-fongo-com-talk-freely-world-free ... #p27825493

Other solutions will typically involve ongoing monthly and/or per minute fees.
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
Jan 18, 2017
36 posts
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i'm having problem with broken incoming voice with obi202 and obi200 while outgoing voice is perfect(in contrast, i'm having broken outgoing voice with HT502 but perfect incoming voice, both are behind router set SmartRG505 with all necessary enabled voip settings from this modem). as i don't want use HT502, i have tried all possible settings from above guide 1.36 but no luck. any idea?
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alienware wrote:
May 30th, 2017 1:02 pm
i'm having problem with broken incoming voice with obi202 and obi200 while outgoing voice is perfect(in contrast, i'm having broken outgoing voice with HT502 but perfect incoming voice, both are behind router set SmartRG505 with all necessary enabled voip settings from this modem). as i don't want use HT502, i have tried all possible settings from above guide 1.36 but no luck. any idea?
You typically can't fix jitter and packet loss with ATA settings. Those are router (QoS) and ISP issues.

Large amounts of jitter produce choppy audio.

High packet loss produces dropped calls.

High pings produce lag or delay.


From page 2 of the PDF guide:

4. It's best to have a decent router for VoIP with strong QoS features.
Stick your ISP's modem in bridge mode, use your own router, and properly enable QoS for your ATA (if you’re going to use adaptive QoS, give your ATA the highest priority for internet traffic and assign lower priorities for all other devices on your LAN). Refer to your router's manual.

I'm not a big fan of this site, but for a general QoS description, visit http://www.voipmechanic.com/qos-for-voip.htm (avoid anything this site says about the G.729 codec because you really don’t want to be using this low bitrate codec unless you’re using Freephoneline on a smartphone with a poor cellular data signal).


from page 39


I'm getting choppy audio, what should I do?


You’re experiencing jitter.

Generally speaking it's best to have a decent router for VoIP with strong QoS features.
Stick your ISP's modem in bridge mode, use your own router, and properly enable QoS for your ATA. Refer to your router's manual or contact your ISP if your router was provided by your ISP.

I'm not a big fan of this site, but for a general QoS description, visit http://www.voipmechanic.com/qos-for-voip.htm (avoid anything it says about G729 codec).
When you test, pick the location that is closest to your VoIP service provider's server location.

1) The typical reaction would be to try enabling QoS properly in your router for your ATA. Refer to your router's manual.

2) Another possibility is you're dealing with congestion during prime time (8p.m. to 11 p.m., especially on Sundays). That's an ISP issue (possibly oversold its service in your area). With Rogers or a cable ISP, you could very well be dealing with local node congestion.

Try running http://vac.visualware.com/ at 8p.m. (especially on a Sunday).

After visiting the link, choose a test location that’s closest to server (Freephoneline’s servers, at this time, are in southern Ontario) you’re using. A MOS score below 4.0 is bad news. It means call quality will not be good. The advanced (+) tab provides interesting info.

You should also try the winmtr test I mention in step 7c ii of the setup guide around 8 p.m. to the server you're using. (scroll down to the bottom section of this post)

If the problem only occurs during prime time (as opposed to weekday mornings), then I would probably start thinking your ISP is to blame.

3) Another possibility is that your ISP uses poor routing tables to Freephoneline’s SIP servers. This usually causes pings to increase (not necessarily jitter). Large ISPs typically won’t do anything if you complain. Smaller ones might.


But no other devices are being used when I experience choppy audio!

A lot of people say that without realizing other devices and/or programs may actually be using bandwidth in the background. It's really not a good idea, in general, to be using a router that doesn't have a good QoS feature for VoIP.

But if what you claim is really true, then you may be dealing the possibility of congestion during prime time (8p.m. to 11 p.m., especially on Sundays). That's an ISP issue (possibly oversold its service in your area/local node congestion).

You should also try the winmtr test (or if you're on a MAC, maybe this helps): https://www.reddit.com/r/TagPro/comment ... tr_on_mac/

Again, if the problem only occurs during prime time (as opposed to weekday mornings) and especially Sunday evenings, then I would probably start thinking your ISP is to blame. Sunday evening is when everyone in your neighbourhood is home.



---

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

Use winmtr: http://winmtr.net/download-winmtr/. Ping about 100 times to each server.


My pings to

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


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

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

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

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

I get between 11 (voip.freephoneline.ca and voip2.freephoneline.ca)-24ms (voip4.freephonline.ca) on average, depending on the server I'm testing to. Preferably, you want pings below 100ms.

Anything over 200ms is unacceptable.

What you don't want to see is 40, 45, 50, 35, 500, 40, 30, 45, 700. That's bad jitter.
You want relatively consistent pings without a lot of variation.

One reason why jitter can occur is due to other devices on your LAN (local area network) using bandwidth. That’s why properly enabling QoS in your router for your ATA is always a good idea. Refer to point 4 in the Preamble.

Bad jitter can produce broken up audio or choppiness during phone calls. Severe jitter can cause calls to drop. Ping affects delay.

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

Ping is a measurement of data packet transmission, and ping does affect delay or lag. All gamers know, almost inherently, that lag affects them negatively. A PC gamer will pound his or her keyboard in hope that a character will respond on his or her monitor, quickly, but when there's a delay or lag, reality doesn't meet expectation. A gamer can see this problem visually. Over VoIP, anything over 200-210 ms, you will typically start to encounter crosstalk due to increased delay, even if the untrained ear doesn't notice. All VoIP services are subject to the same scientific principles including the fact that speed of transmission affects delay, and Freephoneline is not some magical service that is somehow exempt from issues arising from high pings and jitter. When pings and, especially, jitter are high, it's a pretty horrible experience, just as it would be with any other VoIP service. When pings and jitter are fine, Freephoneline is great.

Lastly, anyone using any communication service (or even when playing online games or using other online services) should understand that the longer the path to the server being used, the greater the potential exists for a problem to occur somewhere along that path. Freephoneline’s SIP servers are located in southern Ontario.
Last edited by Guest1284983 on May 30th, 2017 7:34 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.
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Jul 26, 2005
3576 posts
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Montreal
Is there any reason why my OBI tells me that my freephoneline account is not active suddenly? I'm seeing in the callogs a status of "Destination out of order" - timestamps seem to match with a couple of hours following a power outage that I had at home..

EDIT: I solved the problem. Seems like after the power failure, the obi202 was not able to obtain in IP address. a reset of the nano router and the obi202 and it was all fixed.

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