Fongo ATA $49

Webslinger wrote: ↑

scoobydoo wrote: ↑
"magicJack does not charge anyone for inbound telephone calls."

If your friend uses a data plan, and for whatever reason it charges them money to call you with the Magicjack app, all they need to do is find a source of free wifi. Luckily, for my friends in South Korea, it's everywhere.

That is no different than practically every other VoIP service out there, including Whatsapp, Skype, Fongo Mobile, etc.

I've already asked this before and people said it could be not done,

but you're saying even if someone doesn't have a Fongo adapter/account that they can call me FREE from any country by using the Fongo app after downloading it from any online app store?

scoobydoo wrote: ↑

Webslinger wrote: ↑

scoobydoo wrote: ↑
"magicJack does not charge anyone for inbound telephone calls."

If your friend uses a data plan, and for whatever reason it charges them money to call you with the Magicjack app, all they need to do is find a source of free wifi. Luckily, for my friends in South Korea, it's everywhere.

That is no different than practically every other VoIP service out there, including Whatsapp, Skype, Fongo Mobile, etc.

I've already asked this before and people said it could be not done,

but you're saying even if someone doesn't have a Fongo adapter/account that they can call me FREE from any country by using the Fongo app after downloading it from any online app store?

Yes, they can call you for free right now, even if you don't have a Fongo account (with the understanding that they must be able to download Fongo Mobile and register an account) provided you have a phone number in one of these cities: https://www.fongo.com/calling/free/

They don't have to pay for anything. Fongo Mobile is free to download and use.

But as I mentioned before, they're going to be connecting to Fongo's SIP servers in Ontario.

AFAIK, Magic Jack doesn't have servers in South Korea, either: https://en.wikibooks.org/wiki/MagicJack ... roxy_Find2
If so, as a solution for someone in South Korea, I wouldn't recommend using MJ either.

I bought the obi202 and used the anveo service. However the outgoing call name wont work, only displayed the number. That's the only issue i had for about two months. Anyone know how to fix this? Did some rearch and looks like the setting is coming from the obi device, I have not found any solution on google yet.

kumo wrote: ↑ I bought the obi202 and used the anveo service. However the outgoing call name wont work, only displayed the number. That's the only issue i had for about two months. Anyone know how to fix this? Did some rearch and looks like the setting is coming from the obi device, I have not found any solution on google yet.

1) If you used the Obitalk web portal (www.obitalk.com) to configure your ATA, keep in mind that you must continue using it to configure your ATA. Otherwise whatever settings you change will eventually be overwritten by what you previously entered at obitalk.com anyway. If you wish to disable this behaviour, dial ***1. Enter that IP address into a web browser. Navigate to System Management-->OBiTalk Provisioning-->select Disabled for the method. Save. Reboot ATA. Now obitalk.com won't overwrite whatever changes you make via the device's interface (via IP address).

Pick one method (obitalk.com) or the other (IP address of device) for changing device settings. But do not use both methods. Keep in mind that activating Google Voice requires using the Obitalk.com web portal.




That grey cog wheel with the "E" is for the expert configuration menu.

It shows when logging in at http://www.obitalk.com, selecting "Edit Profile" on the left, then scrolling down
under "Advanced Options" and finally selecting "Enable OBi Expert Entry from Dashboard."


2) Navigate to Voice Services-->SP(Anveo) Service-->Calling Features-->CallerIDName

3) Uncheck the "default" box on the right hand side, and if need be, also uncheck the "OBiTALK Settings" box (if you're logged into Obitalk.com)

4) Enter your CallerIDName

5) Navigate to Voice Services-->SP(Anveo) Service-->Calling Features-->AnonymousCallEnable

Ensure that the box under the "Value" column is unchecked. In other words, make sure that AnonymousCallEnable is disabled.

6) Submit/save/reboot ATA

Hi All, I have a teksavvy tektalk configured grandstream ht502, i cancelled the service as I had no need for a home phone. Can i reconfigure this grandstream and use it with a free/super low cost service. I paid $ 50 + tax + shipping for the grandstream and would like to use it for something worthwile...

before anyone suggests I did try selling the grandstream, but there were no takers, so might as well try and use it...

cwazyy wrote: ↑ Hi All, I have a teksavvy tektalk configured grandstream ht502, i cancelled the service as I had no need for a home phone. Can i reconfigure this grandstream and use it with a free/super low cost service. I paid $ 50 + tax + shipping for the grandstream and would like to use it for something worthwile...

before anyone suggests I did try selling the grandstream, but there were no takers, so might as well try and use it...

Provided your ATA is unlocked, you can use any of the services listed here with the exception of Ooma:
newegg-obihai-obi200-ata-49-99-1-50-ehf ... #p26724364
And I recommend doing the ping/jitter tests I mentioned in that post as well.

Webslinger wrote: ↑

1. Are you using FPL on your iPhone regularly? Doesn't UDP burn out your battery faster? Assuming you used TCP before. Is there any benefits of using UDP over TCP?

2. Softphones: I could only find this in the App Store: Acrobits Softphone - SIP phone for VoIP calls by Acrobits
https://appsto.re/ca/FoJUs.i

Is this the same as Groundwire? *edit* found it ( Groundwire - Business Caliber SIP Phone by Acrobits
https://appsto.re/ca/P-3Jw.i )

3. I'm using Media5-fone as I've found issues with Zoiper. Have you tried Media5-fone? *edit* if you've used Media5-fone, how does it also compare?

TheHutt wrote:
1. Are you using FPL on your iPhone regularly?

Not frequently. I use it occasionally.

a) FPL is intended as a home phone replacement, and that's predominately what I use it for. All phone jacks in a house can use VoIP services.

b) Instead of relying on flaky cellular data, which also tends to be more expensive (I'd rather use my data for other things) in Canada than regular cellular airtime, I call into my Obihai ATA or have it call me back to access its services. So, it is possible to use Google Hangouts/Voice, for example, without using cellular data on any cellular phone.

Relying on wi-fi and, in particular, cellular data on smartphones is less reliable.

If you've ever been on a stretch of road where you suddenly only get a weak 3G signal, having to switch to G.729a (a low bitrate, lossy audio codec, which sounds pretty badly under normal conditions anyway) to make a call is not convenient.

Doesn't UDP burn out your battery faster?

Sure, yes, it can, depending on the number of INVITE requests used to keep the connection alive. The issue here is mostly for incoming calls. In some cases, sending frequent keep alive and registration requests is not necessary to keep a connection open (it's probably going to depend on the router and device being used). In my testing on cellular data on Telus' network with FPL using Groundwire, I didn't need to have keep-alive on. And FPL uses a registration timeout of 1 hour anyway.

If constant UDP keep-alive requests keep having to be sent to keep a connection open vs. the default once every two hours with TCP on Android (iirc), then yeah, UDP is going to increase battery drain.


But, Acrobits came up with another method for Groundwire: if you want to save battery life with iOS, use Push notifications: https://www.acrobits.net/hesk/knowledge ... article=52

Assuming you used TCP before. Is there any benefits of using UDP over TCP?

UDP is arguably faster since it doesn't, basically, deal with worrying whether a packet was received by the destination. It used to be argued that UDP was better for congestion. Having 1% packet loss isn't going to ruin an audio stream, for example, whereas with TCP the lost packet needs to be resent (and adds additional time for transmission). So, traditionally live audio streaming was mostly done via UDP. And it still is: with VoIP SIP services, you can only change SIP signalling to TCP. The actual RTP audio stream is always UDP.

Anyway, if a device or app is waiting for packets to be resent (TCP), that does introduce the potential for lag. And in my example where you've got a weak 3G signal somewhere and are dropping packets at a higher rate, TCP (traditionally) isn't going to be much fun.

UDP doesn't have error control. It just spews packets out. There no need for confirmation to be sent to the originating server that a packet was received. There's no negotiating/handshake (ACK/SYN). So, if you've got an unreliable internet service (weak 3G signal), UDP, arguably, has an advantage. But with VoIP, TCP is just for SIP signalling. UDP is still used for the audio stream.

[Edit]: Also, traditionally, there's a greater issue when dealing with TCP and handoffs between networks where IP addresses change. There have been advances though (MPTCP, for example).

3. I'm using Media5-fone as I've found issues with Zoiper. Have you tried Media5-fone? *edit* if you've used Media5-fone, how does it also compare?

Yeah, I wasn't a big fan. I'm not sure how well it works on Android, but back when I was testing on iOS, it would drop calls when handing off between wi-fi and cellular data networks when using UDP transport. On iOS, Groundwire doesn't do that (I'm not sure about Groundwire on Android).

And maybe I missed them (I deleted the app quickly), but I didn't see a record option nor video codec support (but I only tried using that was a business video conference server with Groundwire) at the time, but I didn't look very hard either.

Webslinger wrote: ↑

scoobydoo wrote: ↑

Webslinger wrote: ↑

That is no different than practically every other VoIP service out there, including Whatsapp, Skype, Fongo Mobile, etc.

I've already asked this before and people said it could be not done,

but you're saying even if someone doesn't have a Fongo adapter/account that they can call me FREE from any country by using the Fongo app after downloading it from any online app store?

Yes, they can call you for free right now, even if you don't have a Fongo account (with the understanding that they must be able to download Fongo Mobile and register an account) provided you have a phone number in one of these cities: https://www.fongo.com/calling/free/

They don't have to pay for anything. Fongo Mobile is free to download and use.

But as I mentioned before, they're going to be connecting to Fongo's SIP servers in Ontario.

AFAIK, Magic Jack doesn't have servers in South Korea, either: https://en.wikibooks.org/wiki/MagicJack ... roxy_Find2
If so, as a solution for someone in South Korea, I wouldn't recommend using MJ either.

I don't think it really matters that Magicjack doesn't have servers in South Korea, my ex-girlfriend called me all the time (to Canada) from work and we never had any problems with the call. Thanks for the information and research!

scoobydoo wrote: ↑
I don't think it really matters that Magicjack doesn't have servers in South Korea

It always matters, regardless of the VoIP service being used. As pings increase above 200-210ms, crosstalk will begin to occur, even if the untrained ear doesn't notice.
Related post and video can be found here: newegg-obihai-obi200-ata-49-99-1-50-ehf ... #p26724364

Thanks a lot Webslinger, you are awesome. You fix my issue in a blink. I spent lots of time searching without success.

Here is my ping to freephone line, are these good numbers? I plan to port my home phone to FPL if it's good.

Ping to voip.freephoneline.ca
|------------------------------------------------------------------------------------------|
| WinMTR statistics |
| Host - % | Sent | Recv | Best | Avrg | Wrst | Last |
|---------------------------------------------- --|------|------|------|------|------|------|
| router.asus.com - 1 | 159 | 158 | 0 | 1 | 9 | 1 |
| lo0-0-lns02-tor2.teksavvy.com - 0 | 163 | 163 | 10 | 16 | 69 | 12 |
| ae0-2150-bdr01-tor.teksavvy.com - 0 | 163 | 163 | 10 | 12 | 35 | 11 |
| 295ca.ip4.torontointernetxchange.net - 0 | 163 | 163 | 11 | 13 | 39 | 11 |
| No response from host - 100 | 33 | 0 | 0 | 0 | 0 | 0 |
| No response from host - 100 | 33 | 0 | 0 | 0 | 0 | 0 |
| db.fongo.com - 1 | 159 | 158 | 0 | 12 | 25 | 12 |
|________________________________________________|______|______|______|______|______|______|
WinMTR v0.92 GPL V2 by Appnor MSP - Fully Managed Hosting & Cloud Provider

Ping to voip2.freephoneline.ca
|------------------------------------------------------------------------------------------|
| WinMTR statistics |
| Host - % | Sent | Recv | Best | Avrg | Wrst | Last |
|------------------------------------------------------------------------------|------|------|------|------|------|------|
| router.asus.com - 0 | 104 | 104 | 0 | 1 | 4 | 2 |
| lo0-0-lns02-tor2.teksavvy.com - 0 | 104 | 104 | 10 | 15 | 54 | 14 |
| ae0-2150-bdr01-tor.teksavvy.com - 0 | 104 | 104 | 10 | 13 | 61 | 12 |
| 295ca.ip4.torontointernetxchange.net - 0 | 104 | 104 | 11 | 13 | 40 | 12 |
| No response from host - 100 | 21 | 0 | 0 | 0 | 0 | 0 |
| No response from host - 100 | 21 | 0 | 0 | 0 | 0 | 0 |
| host-162-213-111-22.static.295.ca - 0 | 104 | 104 | 11 | 13 | 37 | 14 |
|________________________________________________|______|______|______|______|______|______|
WinMTR v0.92 GPL V2 by Appnor MSP - Fully Managed Hosting & Cloud Provider

kumo wrote: ↑ Thanks a lot Webslinger, you are awesome. You fix my issue in a blink. I spent lots of time searching without success.

Yeah, these pings are nice a low. The jitter is not significant. You did have 1% packet loss to voip.freephoneline.ca, but that's not a big deal.

Teksavvy in Toronto has always had good routing to FPL's servers.

It's better to test between 8 p.m. and 11 p.m. though (especially on Sundays).

Problems will come up more when you get to your router, make sure your router is able to disable SIP ALG or is disabled by default.

Else you will get all sort of problems!

kumo wrote: ↑ | router.asus.com - 1 | 159 | 158 | 0 | 1 | 9 | 1 |

DealCanuck wrote: ↑ Problems will come up more when you get to your router, make sure your router is able to disable SIP ALG or is disabled by default.

That's not a problem since Kumo is using an Asus router. SIP passthrough (SIP ALG) in Asus routers (using AsusWRT) works fine (enabled or disabled) with FPL and Obihai ATAs.

But SIP ALG is a problem with a lot of other routers.

The SIP passthrough is disabled on my Asus router. It's running the stock firmware not AsusWRT, hope it will work as well.