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[Newegg] Obihai OBi200 ATA $49.99 + $1.50 EHF+ free shipping+tax

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  • Nov 28th, 2017 2:36 pm
[OP]
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Mar 3, 2002
8179 posts
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[Newegg] Obihai OBi200 ATA $49.99 + $1.50 EHF+ free shipping+tax

Remember to use your favourite rebates site before purchasing. This is the cheapest price it's been since Boxing Week 2016.
It was also previously on sale at this price on Nov. 1st: https://forums.redflagdeals.com/newegg- ... g-2138620/.

The RFD referral link keeps redirecting to the wrong webpage. This is the product url:

Code: Select all

https://www.newegg.ca/Product/Product.aspx?Item=N82E16833617008
This Obihai ATA (Analogue Telephone Adapter) does not come with a VoIP service. You provision a VoIP service on the ATA yourself.
The purpose of an ATA is to allow you to use VoIP services with a regular telephone. An Obihai ATA happens to be the most powerful
ATA for home consumer use due to its powerful routing features. Keep in mind, however, that Obihai's support policies are less than desirable, in my opinion.

Image
Image




(OBi202 is shown in video; connection is the same for both OBi200 and OBi202)


Just as an example, in Ontario, the price break down would look like this:
Subtotal: $49.99
EHF: $1.50
GST/HST: $6.70
Shipping: $0.00
--
Grand Total: $58.19


Newegg typically ships via Purolator in Ontario, afaik.


Image (USD pricing)
OBi1xx series no longer supports Google Voice and is also discontinued.
OBi212 replaces the OBi110, but the OBi212 only appears to be sold in the U.S. at the moment: https://www.amazon.com/OBi212-Universal ... B075839DD6.


OBi200

- supports T.38 fax protocol (OBi100 and OBi110 do not)

- supports Google Voice (requires U.S. IP address for Google Voice activation)

- has 1 phone ports

- supports up to 4 different VoIP providers with different phone numbers (plus 8 voice gateways)

- offers Call bridging support/capacity for you to dial into the ATA or have it call you back and allow you to dial through it from a cellphone (this is great for cellular plans with tons of incoming minutes)

- offers USB port (for OBi Bluetooth and OBi wi-fi adapters)

- offers X_AcceptSipFromRegistrarOnly to accept inbound SIP requests only if they came from the same IP address of the current Registered proxy (there are ways to do the same with an OBi100/110, but it's not as simple)

- offers X_EnforceRequestUserID to ensure that the SIP INVITE received by the OBi device has a request userid that matches the SIP account ID (there are ways to do the same with an OBi100/110, but it's not as simple)

- offers X_BlockedCallers for blocking 10 callers easily per Service Provider (you can achieve the same thing using user defined DigitMaps in an OBi100/110--but you're limited to 511 characters per User defined digitmap; so this field lets you add even more phone numbers)




OBi200 lacks (vs. OBi202)

- 2 phone ports (only has 1)

- an internal router (100Mbps in full duplex mode, but the maximum routing throughput between the WAN and the LAN side is approximately 30 Mbps when there are no active calls). OBi100, OBi110, and OBi200 do not offer an internal router




The real benefit of the OBi202 over the 200 is not the router, which most will likely not use, especially when maximum routing throughput between the WAN and the LAN side is approximately 30 Mbps when no calls are active. The real benefit is the extra phone port, which one could use for a dedicated fax machine (or as a separate line).

(from Obihai's latest ad)
Only on the OBi202: Press # for Phone Port Collaboration

Did you know that you can have a mini phone system with the OBi202? While the Phone Port 1 and Phone Port 2 can function independently so you and another person can be on two different calls at the same time, the two phone ports to work together. …Just Press #

- Call the Other Phone – You can press # to call from one phone to the other phone.

- Call Transfer – While on a call, press the hook or Flash button and then press # to ring the other phone. All three of you can talk together or just hang-up to transfer the call to the other phone.

- Join-in on the Other Phone’s Call – If the phone on phone port 1 is on a call, from the phone on phone port 2 press # to join-in on the call.

- Incoming Call Pick-Up – If the phone connected to phone port 1 is ringing, pick-up the phone connected to phone port 2 and press # , then say “Hello?”


Image





OBi200/202 offers support for up to 4 SIP trunks and 8 voice gateways. That means you can use up to 12 different VoIP services on one telephone.

Many people have multiple accounts with different providers. I'm one of them.

For some people, Google Voice is one account. You get a U.S. phone number that people
in a local city in the U.S. can call for free. You can call anywhere in Canada and the U.S. for free.
You can call long distance using GV/Hangouts in CAD: https://www.google.com/voice/rates


Note that obtaining a Google Voice number requires a U.S. IP address and a legitimate U.S. phone number (typically not VoIP) that hasn't been used to activate Google Voice before. I will not be helping people obtain Google Voice phone numbers, sorry.

But if you have a Google Voice account without a Google Voice phone number, you can still provision it on an Obihai ATA to make calls outbound calls to anywhere within Canada and the U.S. (without requiring a U.S. phone number).

Freephoneline is another account. You get a Canadian phone number. You can call to most major Canadian cities for free.
$79.95 setup fee+tax. $25+tax if you want to port your existing phone number into FPL. No ongoing fees for as long as you use FPL.


Someone might be using a VoIP.ms phone number on a SIP account.
https://voip.ms/rates.php (USD)

Someone might be using VoiP.ms on a voice gateway for outgoing calls only.
Some people use free N.Y. phone numbers from Callcentric. That's another SIP account.

Want people to be able to call you from many places around the world for free? http://www.inum.net/?page_id=42
Get a free iNum number from VoIP.ms.
https://wiki.voip.ms/article/Order_a_DI ... ring_iNums



Some people might be using Anveo. Anveo Obihai special pricing can be found here (in USD): http://www.anveo.com/anveoforobitalk.asp
http://www.anveo.com/rates.asp (USD)
If you choose multiple providers, you can cherry pick long distance rates overseas.

You can setup SIP Broker on a voice gateway and get access to free calling to over 2,000 VoIP networks for free:
http://www.sipbroker.com
http://www.obitalk.com/forum/index.php?topic=526.0

Want an incoming phone call to be routed via SIP URI dialing elsewhere for free? You can do that with an OBi. Want to have an incoming call be routed through another service provider to a different phone number? You can do that with an Obihai ATA. Want to be able to dial into your ATA from a cellphone and have the ATA call you back (great if you have free incoming minute plans) and get access to all of the same services you have on your ATA? You can do that with an Obihai ATA's auto attendant feature: https://www.obitalk.com/forum/index.php?topic=66.0

Is your mom is calling you on your FPL number, but you want to route that call to your aunt in the U.S. for free using Google Voice? Or maybe you want that call to be routed to your uncle in Sweden using VoiP.ms' value rates. You can do that with an Obihai ATA.


-----
How do I connect an ATA to my house, so that all existing phone jacks work?

You may need to disconnect your Telco company's line at the demarc--or make sure power from it is not running to your existing phone jacks. Otherwise, you run this risk of frying your ATA. Visit http://www.voipmyhouse.com/#thesolution.

Also, check out bogolisk and canadaodyowner's pictures/posts over here: merged-freephoneline-ca-free-local-soft ... 21229/331/
Better to ask them about it than me.


-----
I've heard scary stuff about Nettalk possibly going bankrupt and people potentially losing their phone numbers until Primus, which appears to be getting taken over, in part, by Birch, obtained Nettalk customers' phone numbers again. What happens if Freephoneline, Fongo, and Ooma go bankrupt? Will I lose my phone number?


Fibernetics, which owns/operates Freephoneline and Fongo, is the largest privately held competitive local exchange carrier (CLEC) in Canada, and FPL/Fongo generates revenue from incoming phone calls or termination fees to its network in addition to the fees paid by its customers. The more phone calls made to their network, the more money they make. Fongo and Freephonline are treated as being separate entities by Fibernetics (it costs money to port phone numbers between FPL and Fongo). So, FPL and Fongo are considered to be sister companies, despite offering similar services.

Fibernetics also operates/owns Nucleus Information Service, Worldline.ca, 1011295.com, 295.ca, Vonix, NEWT, etc.

In the event that Freephoneline were, for some unlikely reason, to suddenly shut down, you would still be able to port out before then:

http://forum.fongo.com/viewtopic.php?f=10&t=16964
"This isn't something you have to worry about. Hypothetically speaking though, if something were to happen, arrangements would be made for users to be able port their numbers out, or stay with whichever company were to take over operations of FPL [or Fongo]. The bottom line is, you would not simply lose your number."--Fongo_Jeff


Nettalk's situation can't happen with FPL or Fongo. Fongo is owned by Fibernetics, which is a CLEC. Fibernetics has FPL's DIDs/phone numbers. If FPL doesn't pay its bills, Fibernetics still has FPL's phone numbers. If Fibernetics doesn't pay its bills, Fibernetics still has FPL's phone numbers/DIDs until another company takes over. And FPL's customers will be able to port out before then or choose to stay with the company that takes over. Moreover, FPL, Fongo, and Fibernetics are registered with CCTS: https://www.ccts-cprst.ca/complaints/service-providers. If there's an issue, you can file a complaint with CCTS, and CCTS will act as an intermediary to help negotiate a resolution for you.

Nettalk is not a CLEC. Nettalk was using Irisitel for its phone numbers/DIDs. Neither are registered with CCTS at the time of this post.
Fongo is young
Fibernetics was founded in 1997.
how sure we are they will be there in 5 year?
No one has a crystal ball. But I'd say Fibernetic's chances are better than Nettalk's.


What happens if Ooma goes under?

Well, unless you can figure out how to unlock the device, you're going to be stuck with a proprietary brick.
You should also check to see if Ooma is listed here (they were at one time):
https://www.ccts-cprst.ca/complaints/service-providers


How can I tell which carrier or CLEC has my phone number?

Enter your phone number here: https://www.twilio.com/lookup


--
I've heard scary stuff about VoIP 911. Isn't it unreliable?

VoIP E911 is a two step process. With Freephonline, after dialing 911, the initial E911 call centre, which does have my name, address, and call back number, still has to transfer the call to local dispatch (PSAP), which doesn't have my name, address, and phone number.

It's important, when signing up to a VoIP service you're planning on using 911 with that you always keep your address updated on file with them. If you move, update your address. Your VoIP service sends that information to the E911 call centre/Northern911, which they will keep on file.

In some rare instances, I suppose it's possible that Northern911 (I'm guessing this is what FPL and other VoIP services in Canada use, but I'm not sure) may not transfer to the correct local dispatch (PSAP) number (human error happens). Some people I configured services for in the past were very paranoid about VoIP E911 and forced me to do a test call. Worked fine. That is, the first person I reached had name and address info; they ask for confirmation. And the call was promptly transferred to local dispatch and correct address info was given to local dispatch, verbally, by the first call centre. Worked fine each and every time I was asked to test.

How does this compare to 911 with a landline?

Landline 911 is not a two-step process. You don't need to keep your address updated. Landlines are the most reliable for 911 calls.
But landlines don't work after your telephone lines have been knocked out by a storm.

How does this compare with Mobile 911?

Mobile 911 is not a two step process. However, they do not have your exact address, but they should have an approximate location (they should at least have the cellular site/tower that's carrying your call), especially if you're in a major city (they may have latitude and longitude). If you're in a rural area, location based on cellular towers may not be very precise. 70%+ of 911 calls are now coming from mobile phones according to the CRTC. Going forward, this is where improvements are going to be made.


Also, keep in mind that with FPL each E911 call is $35. If you dial 911 less than twice a year (or less than every 3 years with Anveo's $1.20 USD/monthly fee) vs. paying $1.50 USD/month with Callcentric or VoIP.ms, you're ahead with FPL. And you're paying an ongoing minimum monthly fee of $3.98 with Ooma. Ask yourself how often you're calling 911. If you're a senior citizen with a lot of health issues, maybe FPL is a bad idea. (And I don't mean to belittle this point. Everyone gets old. Health is a serious matter.) Otherwise, you'll end up way ahead using a FPL in the long run (in terms of cost).

Here's the thing . . . I used to talk to FPL reps several years ago over the phone, back when they allowed tech support calls. And even then a e911 fee was listed (but not in the FAQs), and I inquired about it. I was told the fee was intended to dissuade people from test calling 911--and that people wouldn't actually be charged.

Fast forward to now, and the $35 per call E911 fee is listed in the FAQs. It's listed all over the place. It's certainly enough to prevent me from testing 911 on FPL. Reps are now saying you will be charged no matter what when you dial 911. Is that true? Maybe. Is that enough to scare me from testing 911? Sure. Has anyone been charged yet? I don't know. Anyway, no one is going to be calling 911 using FPL unless it's really necessary now, and if that's the intent, I'm fine with it. And if I really need E911 as a backup (my smartphone is always nearby), it's there for me. In the meantime, I'm not paying ongoing monthly fees for something I'm not using.

freephoneline-ca-free-local-soft-phone- ... #p27964332
__wizard__ wrote:
Jul 4th, 2017 4:42 pm
As a customer with FPL, I used 911 service 3 months ago and never got the $35 charge
YMMV (your mileage may vary)


Obihai OBi200/202 ATAs with the OBiBT adapter can be paired with smartphones over bluetooth: http://www.obihai.com/obibt.
Then with an Obihai OBi 200/202 ATA, you'd add {911:bt} in your OutboundCallRoute, and then all of your 911 calls on your phones go out over your smartphone's 911 cellular service, provided your smartphone remains within bluetooth range of the ATA.


By the way, There's also Anveo's E911 service ($25 USD per year) available through the Obitalk.com web portal, as an alternative 911 service (limited to a maximum of 5 e911 calls per year): https://www.anveo.com/e911obi.asp (click the link for more information). People asking for help with this Anveo E911 service should probably ask canadaodyowner, who is using this service and is also a Freephoneline customer: freephoneline-ca-free-local-soft-phone- ... #p24980477. I have no experience with Anveo's special E911 service.


VoIP E911 is available all the time under these conditions:

1) You have electricity. A UPS is always a good idea.

2) Your internet service isn't out.

3) Your VoIP service isn't down.

I don't know anyone who doesn't have a smartphone.
Last edited by Webslinger on Nov 21st, 2017 2:56 pm, edited 8 times in total.
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
Thread Summary
This Obihai ATA (Analogue Telephone Adapter) does not come with a VoIP service (with the exception of Obitalk, which allows you to call all Obihai devices for free). You provision a VoIP service on the ATA yourself. The purpose of an ATA is to allow you to use VoIP services with a regular telephone. An Obihai ATA happens to be the most powerful ATA for home consumer use due to its powerful routing features. Keep in mind, however, that Obihai's support policies are less than desirable, in my opinion.

This is not a device for people who are unwilling to read and learn.
38 replies
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
8179 posts
2409 upvotes
People should be testing their pings and jitter (you want little to no variation between pings) to the specific VoIP providers' SIP servers they plan on using before purchasing anything.

My pings to

a) voip.freephoneline.ca average 11 ms.
b) voip2.freephoneline.ca average 12 ms
c) voip4.freephoneline.ca average 27 ms

I also see low latency to voip.ms closest sip servers to me as well.
http://wiki.voip.ms/article/Choosing_Server


My pings to VoIP sip servers (FPL, Anveo, voip.ms and to ping.callcentric.com), are well below 50.

Anything over 200ms is unacceptable. You'll begin to encounter crosstalk, even if an untrained ear doesn't notice. So, if you're getting really high pings and jitter, I would avoid the service you're testing.

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.


Ooma is selling a proprietary device and a single service (Ooma's).
Their SIP servers are located in California. They also a SIP server location on the U.S. East Coast now, but I haven't
looked into where that location on the East Coast is. I'm not positive
whether they will offer something more local for Canadian customers in the future.
http://www.monitis.com/traceroute/
208.83.244.94 is one Ooma SIP server that seems to be on the west coast of the U.S.
Regardless, Ooma is a U.S. company, based in California.

Some popular voip services include
freephoneline.ca (servers are in Ontario, I think, possibly around Waterloo, but I'm not positive), voip.ms (wide range of server locations: http://wiki.voip.ms/article/Choosing_Server), anveo.com (Montreal), www.thespout.ca (Vancouver and Seattle), and callcentric.com (New York).




1.Use winmtr http://winmtr.net/download-winmtr/. Ping about 100 times.
When using WINMTR, look at the very last line or hop when checking your pings.

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

When using WinMTR, 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.

Generally speaking it's best to have a decent router for VoIP with strong QoS features.
Stick your ISP's modem/router combo in bridge mode, use your own router, and properly enable QoS in your router for your ATA.



2. For Freephoneline.ca (Ontario based, possibly around Milton or Waterloo), 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.

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, you should probably avoid FPL.

You should also test to make sure FPL works for you before paying anything: https://www.fongo.com/app/desktop/. You will need a microphone and speakers (or preferably a headset).

3. For voip.ms, test to the closest server to you:
http://wiki.voip.ms/article/Choosing_Server

This company has a lot of servers in a lot of different locations.

4. For Anveo (Montreal POP), test to sip.ca.anveo.com

POP= Point of Presence
That's essentially an access point or physical location where two or more types of communication or network devices make a connection.

5. For The Spout (Vancouver), test to ca.sipfrom.thespout.ca
Spout Communications also has servers in Seattle.

Spout Communications can obtain phone numbers for a lot of rural areas in Canada.

Update . . . There may be some issues with Spout: http://www.dslreports.com/forum/r311931 ... ne-call-Or.

6. For Callcentric (New York), test to ping.callcentric.com

7. For Ooma (California), test to myxprov.ooma.com
If I find the east coast server address (or if someone PMs it to me), I'll post it.

By the way, with WinMTR, you should also test to 74.125.39.7 (California) if you plan on trying to use Google Voice.
Obtaining a Google Voice number requires that you have a U.S. IP address and a U.S. phone number first (and I will generally be avoiding questions on how to go about obtaining a GV number).


Pinging and testing to the SIP servers you're thinking of using (and to Google Voice) should always be the first steps before jumping into a voip service.
Do this 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.

If you get horrible results (really high pings and jitter), do not sign up for the service. You will not be happy.
This goes for all VoIP services. Test to their servers first.


This is one reason why I suggest testing first . . .

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 Ooma is not some magical service that is somehow exempt from issues arising from high pings and jitter. I have helped a few people with Ooma. 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, so is Ooma.

Paul's not having jitter issues, but he is experiencing delay:



Start at the 48 minute 20s mark.

https://tinkertry.com/why-i-gave-up-on- ... ne-service
Paul wrote:
"I figured it was time for one last email to Ooma. Can they give me a way to connect my phone calls from a server a lot closer to Connecticut than San Jose, California? It got me a response the next day that gave me a bit of a chuckle


Dear PAUL,

Thank you for contacting Ooma Customer Care. Good day! We are sorry that this isn`t going to work for you. As mentioned before, we only have one server which is in west coast as of yet and we do not have control over with this latency.

If you decide to port your nos. out of Ooma, we will need to keep your account active while the other provider is in the process of porting your nos. out. Please let us know once that is completed so we can remove your nos. from our database.

In case you want to stay with us, we can refund half of the amt. you paid for the Annual Premier.

Please write me back if you have further questions and I will respond to you as quickly as possible.

Thank you for choosing Ooma!

Sincerely,

Ooma Customer Care Specialist
Chat Support is now available 24/7
To reach a live chat agent, please visit us at www.ooma.com/support"

Ooma now has an East Coast server. So Paul may want to test with Ooma again. Back when he was using Ooma, the only server location was in California. However, Ooma may be using the cheapest routing options available after its venture capital ran dry (according to some people), which may make some call quality in some cases less than ideal: http://www.dslreports.com/forum/r30911315-

So go ahead and test.

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. This is one reason why voip.ms is so desirable to some people; it has SIP servers situated in numerous locations (not just in California).


Another factor to keep in mind, is that during prime time (between 8 p.m. and 11 p.m.) cable internet nodes may be oversubscribed 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.

Running http://vac.visualware.com/index.html at 8p.m. (especially on Sunday) may be a better test than a speedtest. Or pick the server that's closest to your VoIP provider's server. A MOS score below 4.0 is bad news. It means call quality will not be good.





What's the easiest way to setup a VoIP service with an Obihai ATA?

BestFind wrote:
Feb 12th, 2016 7:25 pm
I have now received the Obi202 that I bought and have also received the VOIP unlock code from FPL. What now? Do you have step by step instruction onto how to configure my Obi202 with FPL?
I, very quickly, threw together an OBi20x configuration .pdf guide for Freephoneline.
Download link can be found here:
http://forum.fongo.com/viewtopic.php?f= ... 805#p73839.

It's mostly a rough draft. I will proofread it later when I'm feeling a little better.

I just did it for some people I know that need step by step instructions, given the problems with the default settings used on Obitalk.com and also with settings used in the other .pdf guides on FPL's forums.

Hope it helps!


I would recommend posting in this thread instead for FPL issues: http://forums.redflagdeals.com/merged-f ... ip-821229/

The Obitalk portal for initial setup for newcomers may be useful until they get things working the way they want to, especially if they plan on using Google Voice. After you're comfortable and setup properly, stop using the portal (refer to http://forums.redflagdeals.com/newegg-o ... #p27515997 if you want to stop using the portal).

1)The easiest way to setup a VoIP account on an Obihai ATA is to visit http://www.obitalk.com/obinet/
2)Then create an account/register.
3)Then add your Obihai ATA to the portal.
4)Then click the SP# (for FPL use SP1) you wish to configure.
5)Then scroll down the page and select the "next" button for "Obitalk Compatible Service Provider" (you will have to pay for a VoIP service, unless you have a Google Voice phone number).
6)Click "Accept" on the 911 pop up window.
7)Select the appropriate VoIP provider from the list.
8)If you're using your VoIP service for 911 service, then ensure "Use This Service for Emergency 911 Calls" is selected.
10) If required, select the closest Service Provider Proxy Server to you (or chose the one you get the lowest pings/jitter from).
11)Fill in Username and Password (provided by your VoIP provider. With FPL, log in at https://www.freephoneline.ca/showSipSettings; SIP Usename and SIP password).
12) Click "save"


I was reviewing your post and saw that it seems that one can go thru obitalk or IP address but not both.
correct
Since obitalk.com seems to ask for $10 (eventually outside of my warranty period) I would want perhaps to start to do manually.
They only charge $10 USD for updating firmware via the Obitalk web portal outside of your warranty period (and if you to email or call them for technical support outside of the warranty period). That's it. You can still update firmware manually for free.

You can still use the web portal to do whatever you want, after updating firmware, outside of the warranty period.

Can You give me a link that can guide me?
Visit https://www.obitalk.com/info/faq/OBi202 ... b-from-WAN

For FPL manual setup guides . . .

I, very quickly, threw together an OBi20x configuration .pdf guide for Freephoneline.
Download link can be found here:
http://forum.fongo.com/viewtopic.php?f= ... 805#p73839.

It's mostly a rough draft. I will proofread it later when I'm feeling a little better.

I just did it for some people I know that need step by step instructions, given the problems with the default settings used on Obitalk.com and also with settings used in the other .pdf guides on FPL's forums.

Hope it helps!




Penrose wrote:
Jul 5th, 2016 10:40 am
I have the device and would like to get going using it but the only tutorials I can find is using it with google voice account, which I don't have because I am in Canada. are there some tutorials for setting up and using this device that is better suited for Canadians without google voice.?
cessnabmw wrote:
Jul 5th, 2016 11:29 am
What is the preferred method for changes? IP to the Obi or Obitalk.com?
Obitalk.com is required, using post OAuth 2.0 firmware (which your Obihai ATA will come pre-installed with), to provision Google Voice.

I would configure SP1 with with your primary Canadian phone number service provider. SP2 would be GV, if you have it. And if you have no phone numbers with VoIP.ms, then set that up as a Voice Gateway:
http://forums.redflagdeals.com/newegg-o ... #p27516002

After everything is setup, I would stop using Obitalk.com: http://forums.redflagdeals.com/newegg-o ... #p27515997



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.


Image

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."



Service Provisioning Guides

1. Scroll up, and read the steps for Obitalk.com (applies to any provider)

2. Freephoneline (based in Ontario, possibly around Waterloo or Milton):

I, very quickly, threw together an OBi20x configuration .pdf guide for Freephoneline.
Download link can be found here:
http://forum.fongo.com/viewtopic.php?f= ... 805#p73839.

It's mostly a rough draft. I will proofread it later when I'm feeling a little better.

I just did it for some people I know that need step by step instructions, given the problems with the default settings used on Obitalk.com and also with settings used in the other .pdf guides on FPL's forums.

Hope it helps!

(CAD)

3. VoIP.ms (lots of servers all over the place): https://wiki.voip.ms/article/OBi_100/110_%26_OBi_200
Or to setup this service as a voice gateway, visit http://forums.redflagdeals.com/newegg-o ... #p27516002

(USD)

4. Anveo (Montreal POP): http://www.anveo.com/faqobitalk.asp?cod ... lk_general (USD)

5. Callcentric (New York): https://www.callcentric.com/support/dev ... hai/obi202 (USD)

Callcentric offers free New York phone numbers (free incoming calls only): http://www.callcentric.com/dids/free_phone_number

6. Spout Communications (Vancouver and Seattle): http://www.thespout.ca/portal/knowledge ... tails.html
(this really isn't a guide for Obihai ATAs, but you can find basic information there)

I've never used Spout, but dragonc has: http://forums.redflagdeals.com/voip-pro ... #p21776503
So, maybe ask dragonc about Spout Communications if you have questions.

Spout Communications offers phone numbers for a lot of places in Canada that other providers don't.

Prices are in CAD for Spout.

Update . . . There may be some issues with Spout: http://www.dslreports.com/forum/r311931 ... ne-call-Or.

7. You can setup SIP Broker on a voice gateway and get access to free calling to over 2,000 VoIP networks for free:
http://www.sipbroker.com
http://www.obitalk.com/forum/index.php?topic=526.0

8. Want to be able to dial into your ATA from a cellphone and have the ATA call you back (great if you have free incoming minute plans) and get access to all of the same services you have on your ATA? You can do that with an Obihai ATA's auto attendant feature: https://www.obitalk.com/forum/index.php?topic=66.0
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
Member
Oct 7, 2009
364 posts
125 upvotes
Toronto
Have been using this device connected with 2 lines since last 3 years with google voice and freephoneline.ca
Very happy with both, device and the service from freephoneline.
Saves me a huge on landline phone bills + long distance minutes from my cellphone bills.
Last edited by jigz787 on Nov 21st, 2017 2:59 pm, edited 1 time in total.
I am a Public Mobile subscriber
I am a Koodo subscriber
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
8179 posts
2409 upvotes
Obihai support and contact information can be found here: http://www.obihai.com/support


There's two ways to update firmware (for free even if the ATA is out of warranty)--without using the Obitalk.com web portal or paying $10 USD for support when your device is out of warranty:

1. Dialing ***6 (and then pressing "1" if an update is available)

http://www.obihai.com/docs/OBiProvisioningGuide.pdf (page 15)

This is a faster method than using the Obitalk web portal since you don't have to log into anything.
However, lately, even if your ATA is within the warranty period, or even if you've paid $10 USD for extended support, this method
typically lags behind using the Obitalk.com portal and/or doing manual updates yourself.

If this doesn't work despite your firmware not being the latest version, you'll have to try #2.

2. Manually updating firmware via the device:
http://www.obihai.com/docs/OBiDeviceAdminGuide.pdf (page 43)

Also found after visiting http://www.obihai.com/docs-downloads and clicking on "OBi Device Firmware"

In other words, the other firmware update methods are documented both online and in the device manuals.

Firmware history and release notes can be found here: http://www.obitalk.com/forum/index.php?topic=8982.0
(scroll down to the last post). Obihai does not regularly post release notes for the latest firmware versions.


OBi200/202 firmware release notes are found at https://www.ukvoipforums.com/viewtopic. ... =987#p4313. (The latest release notes may not be available).


Firmware 3.2.1 (Build: 5757EX) is the latest official OBiExtras firmware version for both the OBi200 and OBi202: http://fw.obihai.com/OBi202-3-2-1-5757EX.fw.
Note that if you have OBiPLUS firmware installed already on an OBi202 and if your OBiPLUS subscription is expired, updating firmware will completely kill OBiPLUS on the OBi202 forever.



Avoid 3-0-1-4972: http://www.obitalk.com/forum/index.php?topic=10520.0

--


This comes directly from Obihai Support:

a) Will Obihai continue to allow its customers to update firmware manually without using Obitalk.com if their devices are out of warranty and without paying $10 annually? Yes/No

Answer: YES



b) Will customers who do not pay $10 annually still be able to dial ***6 to update firmware without using the Obitalk.com web portal?

Answer: yes

c) Is Obihai planning to completely block customers who do not pay $10 annually from updating device firmware manually and also block manual firmware file downloads?

Answer: We don't do any blockings.


--

Obihai is charging after one year to use their web portal to upgrade firmware (via the web portal)--or for support. That fee has nothing to do with activating or configuring Google Voice, which can still be done, for free, even if the device is out of warranty after one year. And firmware can still be updated manually after one year as well, for free, as well.

Read: http://www.obitalk.com/forum/index.php? ... 2#msg66682

"Regarding the new extended support fee, here are some facts:

The optional fee would cover 1:1 customer support from Obihai for another year, along with firmware updates downloaded and applied via clicking the yellow triangle on the portal site.
Obihai is not locking anyone out from configuring Google Voice via the portal.

Regardless of warranty status, the latest firmware is available for anyone to download and install manually, at no charge, at this post: http://www.obitalk.com/forum/index.php?topic=9.0
Devices which have been manually-updated to the current firmware level can be added to the portal and configured as usual, regardless of warranty status.
The new extended support fee is no different than most manufacturers' policies regarding providing customers with 1:1 support post-warranty, and, as long as you don't need that kind of help, there is no "ransom" or mandatory charge to continue using OBi devices.
See this discussion for instructions: http://www.obitalk.com/forum/index.php?topic=10065.0


Note: if your OBi is currently within its original one-year hardware warranty period, the portal will offer you an optional one-year extended hardware warranty, plus extended customer support (for $20 or $30 US, depending on the device). If your OBi is currently past its original one-year hardware warranty, the portal will offer you optional extended 1:1 direct customer support for $10 instead. The former option would include service for failed hardware (e.g. device replacement), whereas the latter would include just technical support and firmware updating via the portal. The key word here is "optional".

-- SteveInWA (Obihai beta tester)
Last edited by Webslinger on Nov 21st, 2017 3:00 pm, edited 1 time in total.
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
8179 posts
2409 upvotes
T.38 fax protocol works with Freephoneline and an Obi200/202 ATA.

VoIP.ms and Anveo (retail), at the time of this post, only support T.38 fax protocol on the backend via their respective online fax web portals.

If you allow FPL’s voicemail system to answer an incoming fax call, it will be automatically converted to PDF file format. If you have voicemail to email enabled in FPL’s web portal, the PDF will be automatically emailed to you for easy viewing. Login at https://www.freephoneline.ca/voicemailSettings.



For faxing with an OBi200/202 (and FPL),

1) Try firmware 3.2.1 (Build: 5757EX): http://fw.obihai.com/OBi202-3-2-1-5757EX.fw (also works with OBi200). Note that if you have OBiPLUS firmware installed already on an OBi202 and if your OBiPLUS subscription is expired, updating firmware will completely kill OBiPLUS on the OBi202 forever.


OBi200/202 firmware release notes are found at https://www.ukvoipforums.com/viewtopic. ... =987#p4313. (The latest release notes may not be available).

Also visit http://www.obitalk.com/forum/index.php?topic=9.0.

2) 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.


3) Navigate to Codecs-->Codec Profile (A or whatever the VoIP service you're using is assigned to . . . you can determine this under Voice services-->SP[freephoneline] Service-->X_CodecProfile),

4)ensure FAX Event is enabled (this is not a default setting), and

5)ensure under Codec Settings--> that FaxPassThroughCodec is set to G711U

6)T38Enable should be checked

(5, and 6 should be default settings)

7) T38ECM is checked for me (and seems to work). This is not a default setting.


(submit/save)

8. I would increase volume slightly:
Navigate to Physical Interfaces-->Phone Port-->

a) Change ChannelTxGain to -1
b) Change ChannelRxGain to 0


(submit/save/reboot)

9) On your fax machine, lower baud rate to 9600 bps (I'm able to fax at faster rates than 9600, but if you can't without outgoing faxes failing, lower your baud rate to 9600)

10) On your fax machine, turn off or disable ECM (both TX and RX)
http://www.voipmechanic.com/voip-fax-settings.htm


In your call status page, during T.38 protocol fax transmission, you'll see the following:
"Audio Codec = tx=; rx=G711U"
(which doesn't specifically state T.38, but this is the only indication OBi seems to provide)

If you're not using T.38, you'll see "Audio Codec = tx=G711U; rx=G711U" instead, which just means you're using the G.711u codec only.
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
8179 posts
2409 upvotes
Having problems with SIP Scanners? Is your phone ringing constantly with caller ids that appear as 1001, 999, etc. Bots/crackers/scammers are looking (scanning ports) for ways to break into your services and devices.


1. Are you port forwarding from the router to the ATA or using DMZ? Let's not do that unless you have no other choice. Disable any port forwarding in the router to the ATA, especially UDP port 5060. If you find disabling port forwarding creates 1-way audio issues (or other weird problems), try disabling SIP ALG in your router.

2. If you used the OBitalk web portal to configure your ATA, you need to continue using www.obitalk.com for now. Enter the expert menu (advanced configuration; it's an "E" icon). Otherwise, dial ***1, and enter the IP you're told into your web browser.

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.

3. Navigate to Voice Services-->SP(service you're using) Service-->X_UserAgentPort
Change this to something between 30000 and 60000

(In the Obitalk.com Portal, uncheck both device default and obitalk settings boxes to enter in your own settings).

(Submit/save and reboot ATA)

For OBi100 and OBi110

4. Create a white list of authorized IP addresses of the SIP servers you're using (and want to connect with your OBi ATA):
Service Providers>ITSP Profile (service you're using) >SIP>X_AccessList (enter valid SIP server IP addresses).

voip.freephoneline.ca is 208.65.240.44, for example.
voip2.freephoneline.ca is 162.213.111.22
voip4.freephoneline.ca is 162.213.111.21

toronto.voip.ms is 184.75.215.106.

Separate SIP server IP addresses that you use with this ITSP Service profile with commas in X_AccessList. Basically, you need to know what the IP addresses are of the SIP servers you're using for this particular VoIP service (and not for every single VoIP provider you use in general) on this particular ITSP Profile.


(submit/save and reboot ATA)


5. Stick/Add {>('yourauthusernamegoeshere'):ph} in your inbound call route. Voice Services-->SP(service you're using)-->X_InboundCallRoute
Use Oleg's method: http://www.obitalk.com/forum/index.php?topic=5467.0 (step 4 from that link)

If you don't know what yourauthusername is, navigate to Voice Services-->SP(service you're using) -->SIP Credentials-->AuthUserName

Here's an example of what an X_InboundCallRoute might look like with that part added:

{(MTelemarketers):},{>('yourauthusernamegoeshere'):ph}


The first section can be whatever you currently have in X_InboundCallRoute. The bolded part is what you need to add.

(submit/save and reboot ATA)


For OBi200 and OBi202 steps 4 and 5 are a lot simpler:

4. Enable X_AcceptSipFromRegistrarOnly to accept inbound SIP requests only if they came from the same IP address of the current Registered proxy (found under Voice Services > SP(service you're using) Service-->SP Service)
If you're using Callcentric (ITSP service provider) with a secondary registration, don't do step 4 with an OBi200/202.

Note that unless your ATA is registered with voip.freephoneline.ca, Fongo Mobile calls to your Freephoneline phone number will be dropped straight to voicemail if X_AcceptSipFromRegistrarOnly is enabled. Fongo Mobile calls to Freephoneline phone numbers are SIP URI calls.


5. Remember: if you used the OBitalk web portal to configure your ATA, you need to continue using www.obitalk.com for now. Enable X_EnforceRequestUserID to accept SIP invite requests only if the request userid matches AuthUserName or X_ContactUserID (found under Voice Services > SP(service you're using) Service-->SIP Credentials)

(submit/save and reboot ATA)

The combination of steps 4 and 5 will stop sip scanner calls completely. But nothing beats a good firewall.



Having problems with Telemarketers?

For Freephoneline, Follow Me in your Freephoneline web portal must be disabled (unless you route MTelemarketers somewhere where the call is picked up immediately) for call blocking via your Obihai ATA to work. Login at https://www.freephoneline.ca/followMeSettings and check your Follow Me settings.

To learn about MTelemarketers (above) and blocking Telemarketers, visit http://www.toao.net/503-blocking-telema ... an-obi-ata
(this part is unrelated to stopping sip scanners). Good guide. Note that user defined digitmaps are limited to 511 characters.

If you have an OBi200 or OBi202, you can also navigate to Voice Services-->SP (service you're using)-->Calling Features-->X_BlockedCallers
You can enter 10 phone numbers, separated by commas, that you want to block per SP.


A. 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.


B. Navigate to Voice Services-->SP(voipservice) Service-->X_InboundCallRoute

add {(MTelemarketers):}


Here's an example of what an X_InboundCallRoute might look like

Code: Select all

{(MTelemarketers):},{>('yourauthusernamegoeshere'):ph}
For an OBi202, this would look like

Code: Select all

{(MTelemarketers):},{>('yourauthusernamegoeshere'):ph,ph2}
M, by the way, stands for Digit Map.

If you don't know what yourauthusername is, navigate to Voice Services-->SP(voipservice) -->SIP Credentials-->AuthUserName


C. submit/save/reboot

D. Navigate to User Settings-->User Defined Digit Maps

i. Pick an unused User Defined Digit Map

ii. For the Label, enter Telemarketers

iii. For the DigitMap, enter phone numbers you want to block.
For example, (1234567890|4168888888|5193333333)


Some people ask about blocking anonymous or unknown calls
(?|un@@.|Un@@.|anon@.|Anon@.)


? = no Caller ID
@ = any single alphanumeric (number or letter) except #
@ . = zero or more occurrences of any length alphanumeric (number or letter) sequence except # (note that there should be no space after the @ symbol; I had to put it there on this forum)
@@ . = any single alphanumeric (number or letter) except # followed by zero or more occurrences of any length alphanumeric sequence except # (note that there should be no space after the @ symbol; I had to put it there on this forum)

un@@ . will catch unknown or anything starting with un followed by at least one more character (except #)
(note that there should be no space after the @ symbol; I had to put it there on this forum)
anon@. will catch anonymous or anything starting with anon (except #)



I do not generally recommend blocking anonymous calls since doctors and hospitals can call from them.

E. Submit/save/reboot

Note that you must enter phone numbers as they appear in your VoIP service's call log. For FPL users login at https://www.freephoneline.ca/callLogs


Note that this method for Freephoneline drops all Telemarketer calls to FPL's voicemail (FPL basically wants all incoming calls picked up no matter what because FPL makes money off of incoming termination fees to its network), but at least your phones won't ring.

I probably do not have time to troubleshoot the following FPL workaround for that voicemail issue (especially not via PM, thank you), but here's a potential solution for that:

Because of not wanting these telemarketer calls to drop to FPL's voicemail, boon1 came up with a cool idea for sending these calls to the auto attendant:
http://forums.redflagdeals.com/freephon ... #p21660123
However, for me, that's a bit of a problem because people in my household use the Auto Attendant to dial into and receive calls back from (and I don't want them to hear voice prompts that are intended for telemarketers). Because I have an OBi202, I have access to OBiPlus Basic, which gives me access to two additional auto attendants for free. I used one of them: freephoneline-ca-free-local-soft-phone- ... #p21807239 Edit: It appears that OBiPlus has been completely killed by Obihai now.



Alternatively, if you have another ITSP, configured on SP2 for example, you could use

Code: Select all

{(MTelemarketers):sp2(phonenumbertosendtelemarketers)}
in FPL's X_InboundCallRoute in place of {(MTelemarketers):} to send those telemarketing calls to another phone number.

If FPL is SP1, you can also use

Code: Select all

{(MTelemarketers):sp1(phonenumbertosendtelemarketers)}

or (for sip calls)

Code: Select all

{(MTelemarketers):sp1(sipnumber@sipdomain.com)}
It doesn't really matter. But if you don't want telemarketing calls to drop straight to FPL's voicemail, it is possible with an Obihai ATA, to route these calls elsewhere. Maybe you want to send them to Lenny: http://toao.net/595-lenny (keep in mind that sending telemarketers to Lenny will let telemarketers know your phone number is active).

Update

I think this might be a better solution for Telemarketers for FPL users than what I posted previously.

Here are the steps I took:

1. Went to www.tropo.com
2. Created a free developer account
3. Verified account and logged in
4. Found an audio file that plays SIT tones followed by a "We're sorry, you have reached a number that has been disconnected..."
5. Clicked on "My Files" in Tropo and stuck the file in the www folder
6. Selected "My Apps" and clicked "create application"
7. Entered nogood for Basic information (you can put whatever you want here)
8. Clicked on "new script"

Entered the following:

Code: Select all

<?php
say("http://hosting.tropo.com/mytropoaccount#/www/disconnectedmessageaudiofilethatIadded.mp3");
say("http://hosting.tropo.com/mytropoaccount#/www/disconnectedmessageaudiofilethatIadded.mp3");
?>
9. Saved the script as nogood.php (just has to end with .php)

10. Clicked "create app"

11. Scrolled down and picked a free Tropo phone number for Canada

12. Stuck {(MTelemarketers):sp1(mytropophone#)} in X_InboundCallRoute for FPL in my OBi

(where SP1 = FPL), but it doesn't matter what SP you use, as long as you call your Tropo phone number for free using it.

Rebooted


You can also create a White list: http://forums.redflagdeals.com/newegg-o ... st23792781


How do I setup Nomorobo with Freephoneline (or any service provider that doesn't support Nomorobo) with an Obihai ATA?

So, I got Nomorobo working with Freephoneline, and really, with an Obihai ATA and a toll free number from access number from Nomorobo, you can use Nomorobo with any service provider.

Keep in mind, if you have Follow Me enabled in your Freephoneline web portal, that in order for Nomorobo to work properly, your Nomorobo access number has to ring simultaneously (or first if you're using sequential forwarding). Freephoneline doesn't allow toll free phone numbers to be used with Follow Me at the time of this writing. If you're having problems getting Nomorobo working with Follow Me enabled, disable Follow Me in your Freephoneline web portal account.

1. Sign up at https://www.nomorobo.com/

2. Select "Landline/VoIP" for your phone type.

3. Select Vonage as your carrier (or choose one that gives you a toll free nomorobo number). The bolded part is the important step.
I don't know if choosing other carriers will provide you with a toll free Nomorobo phone number, but maybe some will.

4. For phone number, enter your VoIP phone number that you want to protect with Nomorobo

5. Click next (and keep this webpage open)

6. Open another tab in your browser. 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.

7. Navigate to Voice Services-->SP(voip service you want to protect with nomorobo) Service-->X_InboundCallRoute

Yours might look like {(yourcellphonenumber1|yourcellphonenumber2:aa($1)},{(MTelemarketers):},{(MLenny):spx(13475147296@in.callcentric.com)},{ph,spx(1866732xxxx@tollfree.alcazarnetworks.com;ui=$1)}

You just need to add the bolded part.
For OBi202s, you would use {ph,ph2,spx(1866732xxxx@tollfree.alcazarnetworks.com;ui=$1)}

The x in SPx needs to be a number between 1 and 4, where you've setup a SIP service provider. For example, if Freephoneline is setup on SP1, then use sp1.

1866732xxxx is your Nomorobo access phone number. You need to replace the xxxx.

Google Voice is not SIP. GV is XMPP with Obihai ATAs.
So, if you don't have a SIP service, you would need to setup a dummy SIP trunk as shown over here by azrobert:
http://www.obitalk.com/forum/index.php? ... 3#msg68413

You can also use sip.tollfreeproxy.com instead of tollfree.alcazarnetworks.com. I'm not sure, out of all the free SIP tollfree termination services available that don't require registration, which is the most reliable.

8. Save settings/reboot ATA

9. Test your nomorobo number. Go back to the other web browser tab with nomorobo. Select "I'm ready. Call me now." Or choose the test call option.

10. When your phone rings, answer it.

That's it.

If you don’t receive the Nomorobo verification call, you may not have to worry.

For example, I’ve received the following question from another Freephoneline user before:

“I tried following your guide to get Nomorobo working with Freephoneline, but for some reason when I dial number I got from website it rings once and then I get a busy signal (okay, that’s what should happen). But when I click on `I'm ready. Call me now’, my home phone never rings.

I've tried alcazar... Have Follow me - Disabled on FPL

Here is my Inbound route for an OBi202:

{(MTelemarketers):},{ph,ph2,sp1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=$1)}

Any ideas what could be my issue?”


A few thoughts occur to me.

A) Possibly Nomorobo is blocking the outbound call due your phone number not being owned by Vonage (or the provider you selected to obtain the toll free Nomorobo access number). If you don't see the test call in https://www.freephoneline.ca/doGetCallLogs, that might be one explanation. However, that's not a perfectly reliable method for determining what's happening because incoming phone numbers only appear in FPL's call log if the incoming call has been answered by you or FPL's answering machine.

B) If the test call to your FPL number is a SIP URI call from Nomoboro, the test call will never reach you because FPL blocks incoming SIP URI calls unless they're from Fongo Mobile.

C) Possibly FPL doesn't allow incoming Nomorobo verification calls anymore.

D) This is the most important point: it doesn't seem to really matter if you can't receive the Nomorobo verification call. If you don't verify the test call, Nomorobo still picks up the call with "please try your call again" when a telemarketer calls. If the test call is verified, the telemarketer hears this: https://soundcloud.com/nomorobo/nomorob ... er-captcha. I might actually prefer if the test call isn't verified.

Regardless, I suspect if you are seeing "busy" in your Obihai ATA’s call history, then your Nomorobo toll free number is working as intended.

Here's a quick way to test:

a) In your X_InboundCallRoute for Freephoneline, use {ph,ph2,sp1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=3194327596)}

For OBi200 users, use{ph,sp1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=3194327596)}

1866xxxxxxx is your toll free Nomorobo access number. The xxxxxxx needs to be replaced.

b) Using your cellphone, dial your Freephoneline phone number.

The call should be answered within 3 seconds of ringing by Nomorobo ("please try your call again" message is played or something similar), and then call is disconnected by Nomorobo.

c) now, in your X_InboundCallRoute for Freephoneline,
try {ph,ph2,sp1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=yourcellnumber)}

For OBi200 users, try{ph,sp1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=yourcellnumber)}

1866xxxxxxx is your toll free Nomorobo access number. The xxxxxxx needs to be replaced.
Similarly, youcellnumber needs to be replaced with your actual cell phone number.

d) Call your FPL number using your cell phone (or by using another provider other than FPL).

You should see the following in your ATA’s call history (found by logging into ATA with a web browser and navigating to Status–>Call History):
Call Failed (486 Busy Here; SP1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=yourcellnumber)

That's what's supposed to happen. If there's a busy response from Nomorobo, then the call will continue to ring your phone ports (beyond 3 seconds).

If Nomorobo picks up the call, then your phone ports won't ring.

3194327596 is a known marketer.
http://www.obitalk.com/forum/index.php? ... 6#msg58226
(thanks to azrobert)
There's a related thread here: https://www.obitalk.com/forum/index.php?topic=10368.20.
Should you have further questions concerning Nomorobo, you may want to try asking there. You do not, at this time of this writing, need to setup a dummy SIP trunk as described in the first page of that thread.



What happens with Nomorobo is that when the incoming caller ID isn't considered a telemarketer, Nomorobo responds with a busy signal, and your phone ports will still ring. If Nomorobo determines the caller is a telemarketer or robodialer, Nomorobo answers the call, and your phone ports stop ringing: https://nomorobo.zendesk.com/hc/en-us/a ... ks-a-Call-
Nomorobo may take a ring or two before answering the call (your phone will likely ring once per telemarketing call).


Keep in mind that, like Lenny, if Nomorobo answers the call, telemarketers will know your phone number is active and possibly add your phone number to other calling lists. So, in my opinion, using Tropo with SIT tones in combination with Nomorobo is the best solution. When Nomorobo answers the call, make note of the incoming phone number and add it to your MTelemarkers Digit Map.


Good luck!
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
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How do I tell anonymous callers that I don't accept anon calls with Freephoneline?


So . . .

I saw this thread: https://www.dslreports.com/forum/r31674 ... on-message.

I'm not typically a fan of blocking anonymous callers because some of them can be doctors, hospital admins, etc.
But it's not unreasonable to give important people your smartphone number instead and to block anonymous calls with your FPL number.

For those who remember the steps I took with Tropo before to block telemarketers, the following steps are very similar.

Here are the steps I took:

1. Went to www.tropo.com
2. Created a free developer account
3. Verified account and logged in
4. Found this audio file: https://www.dslreports.com/forum/r31680429-
5. Clicked on "My Files" in Tropo and stuck the file in the www folder
6. Selected "My Apps" and clicked "create application"
7. Entered anon for Basic information (you can put whatever you want here)
8. Clicked on "new script"

Entered the following:

Code: Select all

<?php
say("http://hosting.tropo.com/mytropoaccount#/www/blockanonmessageaudiofilethatIadded.wav");
say("http://hosting.tropo.com/mytropoaccount#/www/blockanonmessageaudiofilethatIadded.wav");
hangup();
?>
9. Saved the script as anon.php (just has to end with .php)

10. Scrolled down and picked a free Tropo phone number for Canada

11. Clicked "create app"

12. Stuck {(?|un@@.|Un@@.|anon@.|Anon@.):sp1(mytroponumber)} in X_InboundCallRoute (towards the beginning or left) for FPL in my OBi

(where sp1 is FPL), but it doesn't matter what SP you use, as long as you call your Tropo phone number for free using it.

Rebooted

It does take a good 30 minutes to an hour before new Tropo apps to start working properly with phone numbers.


Anyway, anonymous callers are greeted by out of service SIT tones followed by a message instructing them to reveal their number if they want to reach me (in this example).
I'm not sure if I'm going to use this process for myself yet. But some might be interested.
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
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How do I retrieve Voicemail using the telephone for Freephoneline?
To access your FPL voicemail, dial your FPL phone number from your OBi, using FPL.

OR

Log into https://www.freephoneline.ca/mailbox (at this time, don't delete voicemail from there; you run the risk of triggering a bug:
http://forums.redflagdeals.com/newegg-o ... #p28437035)

OR

Dial a Freephoneline voicemail remote access phone number (useful from your smartphone) followed by your FPL account phone number (starting with 1) + #, followed by your voicemail password + #: http://www.freephoneline.ca/vmAccessNumbers

OR

Let's get voicemail access working by dialing *98. I think that's what most of us are used to.

If you used the OBitalk web portal to configure your ATA, you need to continue using www.obitalk.com for now. Enter the expert menu (advanced configuration; it's an "E" icon). Otherwise, dial ***1, and enter the IP you're told into your web browser.

For these changes, don't copy and paste entire sections. Just add the bolded stuff.

1. Navigate to Physical Interfaces-->PHONE Port(FPL)-->PHONE Port->DigitMap

Yours probably looks something like this:

([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|##|**70(Mli)|**8(Mbt)|**81(Mbt)|**82(Mbt2)|**1(Msp1)|**2(Msp2)|**3(Msp3)|**4(Msp4)|**9(Mpp)|(Mpli))

You need to stick |*98| in there somewhere.

So, change that to

([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|*98|**0|***|#|##|**70(Mli)|**8(Mbt)|**81(Mbt)|**82(Mbt2)|**1(Msp1)|**2(Msp2)|**3(Msp3)|**4(Msp4)|**9(Mpp)|(Mpli))

(submit/save/reboot ATA)

2. Navigate to Service Providers-->ITSP Profile (freephoneline)-->General-->DigitMap

It probably looks something like (1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.)

Add |*98| in there.

So, change that to

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|*98|011xx.|xx.|(Mipd)|[^*#]@@.)

(submit/save/reboot ATA)


Now FPL voicemail should be set to work with *98 when you dial. But there's a problem. In the OBi ATA, *98 is set to Blind Transfer. Let's change that to *99.

3. Navigate to Star Codes-->Star Code Profile (Freephoneline)-->Code28

To Figure out what Star Code Profile you should be using look at Physical Interfaces-->Phone Port (FPL)-->Calling Features-->StarCodeProfile
It's probably set to A. So for step 3, it's probably Star Code Profile A that you need to change.

It will show: *98, Blind Transfer, coll($Bxrn)

Change that to
*99, Blind Transfer, coll($Bxrn)

(submit/save/reboot ATA)




Here's an OBihai Star Code quick reference guide: http://www.obihai.com/docs/OBiFeatureStarCodes.pdf


Blind Transfer is neat by the way: http://www.obitalk.com/forum/index.php?topic=3039.0
Obviously, if you change Blind Transfer to *99, you need to use *99.
How do I retrieve the VM using the telephone for Google Voice?
Dial your GV phone number from your OBi, using GV.

or

Login to https://www.google.com/voice#inbox

or login to https://www.google.com/voice#voicemailsettings and configure GV to email voicemail to you
(there are also other settings there)
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
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T-Bone wrote:
Mar 25th, 2013 3:48 pm
Has anyone enabled the Obi device to switch to the alternate server if the primary one goes down? (ie. switch from voip.freephone.ca to voip2.freephoneline.ca)?

I found a thread related to this about voip.ms, but I though we could do it with fpl

Following these instructions will work for FPL, but you can do something similar for other providers that offer multiple SIP servers. For VoIP.ms, this will work for outgoing calls only (not incoming).
Be advised that if your ATA is up and running without internet access, your ATA may try to register with freephoneline.ca (which is not a SIP server) over and over and over again. Registration will fail. Your internet access needs to be available as soon as your ATA is turned on. So don't do this unless you really know what you're doing.

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.


For OBi200/202

1. Navigate to Router Configuration-->WAN Settings-->Local DNS Records

For OBi100/110

1. Navigate to System Management --> Network Settings-->Local DNS Records

(instructions follow for all models)

2. Pick an unused/blank value. Enter

Code: Select all

voipmychoice.freephoneline.ca={voip.freephoneline.ca:5060,x},{voip2.freephoneline.ca:5060,y},{voip4.freephoneline.ca:6060,z}
3. Change x,y,z to a number between 1 and 3, where the number represents priority (that is, what server you want to register with first before others). x,y, and z must be different numbers.
My pings/jitter with voip.freephoneline.ca tend to be better than those from voip2.freephoneline.ca, which in turn are better than pings/jitter with voip4.freephoneline.ca.
So x, y, and z for me would be 1, 2, and 3, respectively.

Change "mychoice" to your first priority. So in this example, you would enter

Code: Select all

voip.freephoneline.ca={voip.freephoneline.ca:5060,1},{voip2.freephoneline.ca:5060,2},{voip4.freephoneline.ca:6060,3}

(submit/save/reboot)

4. Navigate to Service Providers --> ITSP Profile (FPL) --> SIP
5. For Proxy server, enter voipmychoice.freephoneline.ca (in this example, it's voip.freephoneline.ca). If voip4.freephoneline.ca is voipmychoice.freephoneline.ca (for you), then you will also need to enter 6060 for the ProxyServerPort.
6. Registrar server should be blank
7. enable X_ProxyServerRedundancy
(submit/save/reboot)
Jeff146 wrote:
Apr 1st, 2015 2:25 pm
Perfect thanks, can't really test it though unless the server goes down lol.
A. You could try to use a firewall to block one of the servers.

Or

B. a) Temporarily change voip.freephoneline.ca to voip50.freephoneline.ca

Then can see that you'll register on voip2.freephonline.ca, in my example.

b)Next temporarily change voip.freephoneline.ca and voip2.freephoneline.ca to voip50.freephoneline.ca and voip250.freephoneline.ca.
In this example, you can then see that you'll be registered on voip4.freephoneline.ca

c) Make sure to reset these changes after testing (refer to steps 2 and 3 above)
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
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Did you know you can also record an active call?

1. Dial ***1

2. Enter that IP address into a web browser

3. Navigate to Status-->Call status

4. A call needs to be in progress in order to record. Click "call status" when a call is in progress.

5. Click the record button.

6. A window will eventually popup. Click save.

7. An .au audio file will start downloading onto your computer.
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
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Getting one way audio issues with an OBi200/202 and Freephoneline? Are incoming calls not ringing? Can you not hear one side of the conversation (you can hear the caller, but the caller can't hear you or vice versa)?

These instructions do not address "this account is not valid" messages (you would need to contact FPL/Fongo for that problem).
"If you’re getting an “invalid account” error messages, or if people trying to call you are hearing "invalid account" or a busy signal, please log in to your account online at https://www.freephoneline.ca/followMeSettings and reset your Follow Me settings (or disable it). Please ensure your temporary FPL number is not listed as one of the Follow Me numbers."

If you have calls going straight to voicemail, login at https://www.freephoneline.ca/voicemailSettings and ensure "Rings before voicemail" is greater than 1. Also, check in your ATA to ensure you don't have "Do Not Disturb" enabled. This is found after logging into your ATA or at Obitalk.com under Voice Services-->SP(FPL) Service-->Calling Features-->DoNotDisturbEnable. Ensure there is no checkmark under "Value".
Navigate to Voice Services-->SP (FPL) Service-->Calling Features
a) Ensure DoNotDisturbEnable is unchecked
b) Ensure CallForwardUnconditionalEnable is unchecked
c) Ensure CallForwardOnBusyEnable is unchecked
d) Ensure CallForwardOnNoAnswerEnable is uncheked
e) Ensure AnonymousCallBlockEnable is unchecked


Often the problem is due to RTP packets not reaching the ATA. Common causes involve poorly functioning SIP ALGs (especially true with certain Netgear routers) in routers or NAT firewalls.

Hardware Specific Issues

A. Netgear R7000 routers

Update firmware. Disable SIP ALG in this router. Then reboot modem, router, and ATA in that order. Then test again.

If you have a Netgear R7000 router, you may need to install third party XWRT-Vortex firmware. I recommend doing this anyway to obtain easy access to both UDP Unreplied and UDP Assured timeout settings. Afterwards, turn off the router and the ATA. Turn on the router. Wait for it to be fully up and running (including Wi-Fi). Then turn on the ATA. Download XWRT-Vortex here: http://xvtx.ru/xwrt/download.htm. In your router, navigate to Advanced Settings–>WAN–>NAT Passthrough–>SIP Passthrough. Change SIP Passthrough to “Enabled + NAT helper.” Click “Apply.”

B. Nettis 4422 modem from Carry Telecom (click the "Internet" tab)
http://www.carrytel.ca/support.aspx
Q : DSL - My VoIP phone does not work with Netis 4422 modem.
A : Please download the newest Netis firmware at www.carrytel.ca/download/netis1228.zip. Unzip the netis1228.zip file and update the firmware file netis1228.img for your modem. The new firmware has been tested and working with most of Voip phone providers

C. Asus VLAN

A number of people have been trying to eliminate Bell Hubs from their setups by using this guide: http://blog.ngpixel.com/post/1044497475 ... own-router.
At the time of this guide being written, NAT acceleration must be disabled in this setup in order for SIP services, including Freephoneline, to work properly. In your router, navigate to Advanced Settings-->LAN-->Switch Control-->NAT Acceleration. Select "disable." Click "apply."Then reboot your modem, router (wait for Wi-Fi SSIDs to populate first before rebooting ATA), and your ATA, in that order.

To determine whether you need NAT Acceleration enabled, visit https://routerguide.net/nat-acceleration-on-or-off/. If you do require NAT Acceleration to be enabled, don’t use VLAN with Asus routers.

D. Hitron CGN series gateway modem/router combos (from Rogers, Shaw, or another ISP) or any modem/router combo from any ISP with SIP ALG forced on

If you don’t have your own router, and if you can’t get someone from Rogers or your ISP to disable SIP ALG for you in their modem/router combo, your ATA may need to register with voip4.freephoneline.ca:6060. The purpose of voip4.freephoneline.ca:6060 is to help circumvent faulty SIP ALG features in routers. So, if you’re experiencing one-way audio issues as a result of SIP ALG, this is the SIP server to try. Check to ensure that you can’t disable SIP ALG yourself (refer to point E below).

E. Hitron CGN3ACSMR and CODA-4582 series gateway modem/router combos from Rogers (and possibly other ISPs)
Open your web browser, and login at 192.168.0.1. Default username is cusadmin.
Select the “Basic” tab and disable “SIP ALG.” Click the “save changes” button.




For everyone with one-way audio issues, follow these steps:

i. Before beginning the steps below make sure whatever modem/router combo your ISP gave you is in bridge mode if you are using your own router. Call/contact your ISP if you have to. For Bell Hubs, visit http://forums.redflagdeals.com/please-s ... r-1993629/


1. Disable any and all port forwarding and/or DMZ in your router. Port forwarding creates security issues and can open the door to SIP scanners and hackers. If you're having trouble with SIP scanners and/or telemarketers, visit http://forums.redflagdeals.com/newegg-c ... st24563087

2. 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.

(In Obitalk.com, you will need to enable and enter expert settings to do the following, if you want to use Obitalk.com. You do this by selecting Edit Profile-->Advanced Options-->check Enable OBi Expert Entry from Dashboard-->submit)

Keep in mind too, that if you're using the Obitalk.com web portal, after you submit a new setting, it takes several minutes before Obitalk.com pushes the changes you've made to your ATA. Your ATA should reboot automatically after the changes are submitted.


3. In your Obihai ATA or at Obitalk.com, Navigate to Voice Services-->SP(FPL) Service-->X_UserAgentPort
Pick a random number between 30000 and 60000

(submit/save/reboot)

4. Navigate to Service Providers-->ITSP Profile (FPL)-->SIP

i) ensure X_DiscoverPublicAddress is enabled (it is by default)

ii) enable X_UsePublicAddressInVia (it's not by default)
You will need to uncheck default, device default, and Obitalk settings boxes. Then check the box to enable the feature

(submit/save/reboot ATA)

5. Retest
When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.

Then retest by calling your FPL phone number. If the problem is solved, don't continue.

6. If that doesn't work, you can also try enabling X_DetectALG (Navigate to Service Providers-->ITSP Profile (FPL)-->SIP)

(submit/save/reboot ATA)

7. Retest
When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.

Then retest by calling your FPL phone number. If the problem is solved, don't continue.

8. If that still doesn't work, disable X_DetectALG. And submit/save/reboot ATA.

9. If there are still problems, try disabling the SIP ALG feature in whatever router or modem/router combo it is that you're using:
http://www.obihai.com/faq/sip-alg/calling-out
I'm of the opinion Apple routers don't offer this feature, but you might as well check. If you manage to disable SIP ALG in the router, then retest.

DLINK router users may need to log into the admin page of their router, click the "Advanced" tab and then "Firewall Settings",
navigate to "Application Level Gateway (ALG) Configuration", and uncheck SIP: http://www.support.dlink.com/emulators/ ... dv_dmz.htm

If you received a modem/router combo, from your ISP ask your ISP. It is typically better to stick the modem/router combo from your ISP in bridge mode and use an external router.

See here for an example on how to disable SIP ALG in a router: http://www.obihai.com/faq/sip-alg/disable-alg

Image

Save settings.
Turn off both router and ATA. Turn on router. Wait for router to be fully up and transmitting data. Turn on ATA.
Then retest by calling your FPL phone number. If the problem is solved, don't continue.

10. Try voip4.freephoneline.ca:6060

Refer to the underlined notes in section 14 of the PDF guide located here: http://forum.fongo.com/viewtopic.php?f= ... 805#p73839.
Also refer to the pic on page 21.

Retest. When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.

Then retest by calling your FPL phone number. If the problem is solved, don't continue.

voip4.freephoneline.ca:6060 is a SIP server whose purpose is to help those with SIP ALG issues (can't disable it in the user's router, for example).

So steps #6, #9, and #10 are all related. They are attempts to address a problem created by SIP ALG.


11. Try this at your own risk: use voip3.freephoneline.ca as the proxyserver
Make sure you refer to step 2 again.
voip3.freephoneline.ca is intended for testing purposes only--or for those who receive explicit permission to use it. Using it for an extended period may get your account banned. However, if using voip3.freephoneline.ca does work, you should open up a ticket with support and let them know that you can't get two-way audio any other way: https://support.fongo.com/anonymous_requests/new. Request a "forced registartion" in your ticket.

If no responds to your support ticket, provide the ticket number in a private message to Fongo Support: http://forum.fongo.com/ucp.php?i=pm&mode=compose&u=7852

FPL configures its SIP servers differently than many other VoIP providers.
voip3.freephoneline.ca conforms more to the norm. But using it without permission can get your account banned.
If you'd like to avoid getting your account banned, use Proxyserver voip.freephoneline.ca, voip2.freephoneline.ca, or voip4.freephoneline.ca:6060 instead and skip to step #14.

12. Retest. When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.

Then retest by calling your FPL phone number.

13. If none of that helps, then, unfortunately, you're pretty much stuck with port forwarding your RTP (UDP) port range 16660-16798 from your router to your ATA. For reference, that range can be found under ITSP Profile (FPL)-->RTP. Then look at LocalPortMin and LocalPortMax. RTP packets need to reach your ATA in order for you get incoming audio. Quite often, when the one way audio issue occurs, this is the problem. RTP packets are not reaching your ATA. Ideally, one should not have to port forward in order to achieve proper two-way audio, since port forwarding does create security issues. Port forwarding should only be done when everything else fails.

Refer to the port forwarding section of your router manual to learn how to port forward to your ATA. If a router was given to you by your ISP, call your ISP.

14. Retest. When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.


Then retest by calling your FPL phone number.

15. 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.

A 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.

Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers may be
difficult, if not impossible. Asuswrt-Merlin, third party firmware for Asus routers, does offer easy access to these two
settings, which are found under General–>Tools-->Other settings. In part, for this reason, I tend to use Asus routers
that work with Asuswrt-Merlin. However, my understanding is that third party Tomato firmware has these two settings
as well. So if your router supports Tomato firmware, that may be another option.
The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 10 for UDP
Unreplied Timeout and 117 for UDP Assured Timeout.

16. If all else fails, try posting at http://forum.fongo.com/viewtopic.php?f= ... &start=300 and/or open a support ticket at https://support.fongo.com/anonymous_requests/new.
When creating a ticket, for the issue type select VoIP Unlock Key-->My account inquiry. Ask for a "forced registration."
If no responds to your support ticket, provide the ticket number in a private message to Fongo Support after registering and logging into the forums: http://forum.fongo.com/ucp.php?i=pm&mode=compose&u=7852

When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
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2409 upvotes
How do I stop Obihai from updating firmware without my permission and stop using Obitalk.com?


If you don't disable Obitalk Service (in addition to other settings), Obihai can remotely upgrade firmware on your device without your permission, in the middle of a storm with power flickering, while you're out somewhere. If the power goes out in the middle of a firmware upgrade, you run the risk of wrecking your ATA. Do you want a bricked device? I don't. A UPS might be a good idea, by the way (http://www.newegg.ca/UPS/SubCategory/ID-72. Doesn't have to be from Newegg. Tons of places sell them).

If firmware can be upgraded remotely, Obihai has access to your ATA and can make changes. That's fine if you need technical support. It's not fine if you're into privacy.

After 1 year is up, in order to continue using Obitalk.com to upgrade firmware, you'll need to pay a $10 USD annual fee. However, you don't need to pay if you upgrade firmware manually. Best to learn now how to do it manually, unless, of course, you like paying annual fees.

There's two ways to update firmware for free even if the ATA is out of warranty:

A. Dialing ***6 (and then pressing "1" if an update is available)

http://www.obihai.com/docs/OBiProvisioningGuide.pdf (page 15)

This is a faster method than using the Obitalk web portal since you don't have to log into anything.
If no update is found for you, you'll have to do step B.

B. Manually updating firmware via the device:
http://www.obihai.com/docs/OBiDeviceAdminGuide.pdf (page 43)

Also found after visiting http://www.obihai.com/docs-downloads and clicking on "OBi Device Firmware"


New firmware releases offer new options that sometimes aren't quickly added to the web portal. Consequently, imo, it's best to stop relying on the web portal. You can (and, imo, should) use the Obitalk.com portal for initial setup (and keep in mind that you must use the web portal to activate Google Voice on your ATA), but eventually, people should learn to stop using it, in my opinion--unless, of course, you want to use OBiExtras ($4.99/month USD service). I don't.


To ensure Obihai and a ITSP provider can't, without your permission, update your device's firmware remotely, you must do the following (all three steps):

1. Dial ***1, and enter the IP address you're told into your web browser

2. Navigate to System Management-->Auto Provisioning

a) For Auto Firmware Update ---> ensure method is disabled

b) For ITSP Provisioning ---> change method to disabled (uncheck default box)

c) For OBiTalk Provisioning--->change method to disabled (uncheck default box)
Keep in mind changes made in Obitalk.com web portal will no longer transfer to your device
after this change is made.

(submit)

3. Navigate to Voice Services-->OBiTALK Service-->Obitalk service Settings
a) uncheck enable and uncheck the default box
You will not be able to call other Obihai devices using their respective Obitalk numbers after making this change. Also the OBi Echo test will no longer work if you disable OBiTALK Service.

(submit/reboot)

Credit goes to Pianoguy for teaching me about this issue.
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
[OP]
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Mar 3, 2002
8179 posts
2409 upvotes
How Do I Setup VoIP.ms (or another service) as a Voice Gateway? (for outgoing calls only)

@puska83,

You were asking me about voip.ms in PMs, and I brought up setting up this service as a Voice Gateway instead of using up a SP trunk slot.

So, I'm going to respond here in case anyone else wants to learn.

As some of you may be aware, you don't necessarily need to pay for a phone number or DID with VoIP.ms. Perhaps you already have a Canadian phone number from Freephoneline and don't need another one. You don't need to take up a SP slot, in this case.

puska83 has FPL setup on SP2. This is important to keep track of. For a Voice Gateway to work, another SIP Trunk must be defined and established. Google Voice doesn't count because it's XMPP. Keep this in mind when you get to step 1e.

So, I would suggest doing something like this:

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.

(In Obitalk.com, you will need to enable and enter expert settings to do the following, if you want to use Obitalk.com.)


1.
a. Navigate to Voice Services-->Gateway and Trunk Groups
b. Select an unused Voice Gateway
c. Enable check
d. Name VoIP.ms
e. AccessNumber is SP2(toronto.voip.ms) or SP2(whatever sever you have specified in your voip.ms web portal)
d. DigitMap (<*2:>XX.)
You can change *2 to whatever you want. So, when you dial *2 before the phone number, it goes out over VoIP.ms
XX. stands for any number you can punch in.

XX. is an indefinite variable, which basically stands for anything you could possibly dial.

I'm just explaining this now for when you get to step 2a below.

e. AuthUserID your voip.ms userid
f. AuthPassword your voip.ms password

(submit/save/reboot)

2. Navigate to Physical Interfaces-->Phone Port-->DigitMap

a. Add *2XX.S3 to your digitmap (this entry needs to be separated by "|", so use |*2XX.S3|)

Again, you just need to match *2 with whatever you used in step 1d.


((Mop)|[1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**8(Mbt)|**9(Mpp)|(Mpli)|*98|310xxxx|1xxxxxxxxxx|[2-9]xxxxxxxxx|*2XX.S3|**0|***|222222222|**9(Mpp))

Don't copy and paste this. You just need to add the bolded section. That's not even what mine looks like. I'm just providing an example.

XX. is an indefinite variable, which basically stands for anything you could possibly dial.
Because it's an indefinite variable, it's also subject to a 10s interdigit timeout, unless you specify the number of seconds you want your OBi ATA to wait for you to finish punching in a number.

So, XX.S3 would mean there's a 3 second delay before the phone number is sent by your ATA.

b. Navigate to Physical Interfaces-->PHONE Port-->OutboundCallRoute

c. You need to add

{(Mvgx):vgx}

x = the # of the voice gateway you chose in step 1b.
So change x to the number of the voice gateway you chose in step 1b.
M = digitmap

The outboundcallroute is processed from left to right.

So, if FPL is setup on SP2 and GV is setup on SP1, you might have something that looks like this:

{([1-9]x?*(Mpli)):pp},{(<#:>):ph2},{**0:aa},{***:aa2},{(<**2:>(Msp2)):sp2},{(<**1:>(Msp1)):sp1},{(Mvg1):vg1},{(<**3:>(Msp3)):sp3},{(<**4:>(Msp4)):sp4},{(<**8:>(Mbt)):bt},{(<**9:>(Mpp)):pp},{(Mpli):pli},{011xx.:SP2},{911:sp2},{933:sp2},{([1-9]x?*(Mpli)):pp},{(<##:>):li},{(<#:>):ph2},{(<**70:>(Mli)):li},{(<**82:>(Mbt2)):bt2},{(<**81:>(Mbt)):bt},{(<**8:>(Mbt)):bt},{**0:aa},{***:aa2},{(Mpli):pli}

Just need to add what's in bold after whatever you have for sp2 and sp1. Don't copy and paste this in it's entirety. Just look at what I have in bold as an example. And 1 is the number of the voice gateway you chose before. So you may need to change the "1".

d. submit/save/reboot
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
[OP]
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Mar 3, 2002
8179 posts
2409 upvotes
What's the easiest way to setup a VoIP service with an Obihai ATA?

BestFind wrote:
Feb 12th, 2016 7:25 pm
I have now received the Obi202 that I bought and have also received the VOIP unlock code from FPL. What now? Do you have step by step instruction onto how to configure my Obi202 with FPL?
I would recommend posting in this thread instead for FPL issues: http://forums.redflagdeals.com/merged-f ... ip-821229/

I used to recommend using the Obitalk portal for initial setup until newcomers get things working the way they want to, especially if they plan on using Google Voice.
However, I no longer recommend using the Obitalk portal for Freephoneline setup due to settings in the pre-configured profile not being set properly for FPL, in particular, with certain routers. Use this PDF guide for FPL instead: http://forum.fongo.com/viewtopic.php?f= ... 805#p73839.

You will need to use Obitalk.com to activate Google Voice.
After you're comfortable and setup properly, stop using the portal (refer to http://forums.redflagdeals.com/newegg-o ... #p27515997 if you want to stop using the portal).

1)The easiest way to setup a VoIP account on an Obihai ATA is to visit http://www.obitalk.com/obinet/
2)Then create an account/register.
3)Then add your Obihai ATA to the portal.
4)Then click the SP# (for FPL use SP1) you wish to configure.
5)Then scroll down the page and select the "next" button for "Obitalk Compatible Service Provider" (you will have to pay for a VoIP service, unless you have a Google Voice phone number).
6)Click "Accept" on the 911 pop up window.
7)Select the appropriate VoIP provider from the list.
8)If you're using your VoIP service for 911 service, then ensure "Use This Service for Emergency 911 Calls" is selected.
10) If required, select the closest Service Provider Proxy Server to you (or chose the one you get the lowest pings/jitter from).
11)Fill in Username and Password (provided by your VoIP provider. With FPL, for example, log in at https://www.freephoneline.ca/showSipSettings; SIP Usename and SIP password).
12) Click "save"


I was reviewing your post and saw that it seems that one can go thru obitalk or IP address but not both.
correct
Since obitalk.com seems to ask for $10 (eventually outside of my warranty period) I would want perhaps to start to do manually.
They only charge $10 USD for updating firmware via the Obitalk web portal outside of your warranty period (and if you to email or call them for technical support outside of the warranty period). That's it. You can still update firmware manually for free.

You can still use the web portal to do whatever you want, after updating firmware, outside of the warranty period.

Can You give me a link that can guide me?
Visit https://www.obitalk.com/info/faq/OBi202 ... b-from-WAN

For FPL manual setup guides . . .
http://forum.fongo.com/viewtopic.php?f= ... 805#p73839
(For manual setup)



Penrose wrote:
Jul 5th, 2016 10:40 am
I have the device and would like to get going using it but the only tutorials I can find is using it with google voice account, which I don't have because I am in Canada. are there some tutorials for setting up and using this device that is better suited for Canadians without google voice.?
cessnabmw wrote:
Jul 5th, 2016 11:29 am
What is the preferred method for changes? IP to the Obi or Obitalk.com?
Obitalk.com is required, using post OAuth 2.0 firmware (which your Obihai ATA will come pre-installed with), to provision Google Voice.

I would configure SP1 with with your primary Canadian phone number service provider. SP2 would be GV, if you have it. And if you have no phone numbers with VoIP.ms, then set that up as a Voice Gateway:
http://forums.redflagdeals.com/newegg-o ... #p27516002

After everything is setup, I would stop using Obitalk.com: http://forums.redflagdeals.com/newegg-o ... #p27515997



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.


Image

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."



Service Provisioning Guides

1. Click http://forums.redflagdeals.com/newegg-o ... #p27516013 (applies to any provider)

2. Freephoneline (based in Ontario, possibly around Waterloo or Milton): http://forum.fongo.com/viewtopic.php?f= ... 805#p73839

(CAD)

3. VoIP.ms (lots of servers all over the place): https://wiki.voip.ms/article/OBi_100/110_%26_OBi_200
Or to setup this service as a voice gateway, visit http://forums.redflagdeals.com/newegg-o ... st26400985

(USD)

4. Anveo (Montreal POP): http://www.anveo.com/faqobitalk.asp?cod ... lk_general (USD)

5. Callcentric (New York): https://www.callcentric.com/support/dev ... hai/obi202 (USD)

Callcentric offers free New York phone numbers (free incoming calls only): http://www.callcentric.com/dids/free_phone_number

6. Spout Communications (Vancouver and Seattle): http://www.thespout.ca/portal/knowledge ... tails.html
(this really isn't a guide for Obihai ATAs, but you can find basic information there)

I've never used Spout, but dragonc has: http://forums.redflagdeals.com/voip-pro ... #p21776503
So, maybe ask dragonc about Spout Communications if you have questions.

Spout Communications offers phone numbers for a lot of places in Canada that other providers don't.

Prices are in CAD for Spout.

Update: There may be some issues with Spout! http://www.dslreports.com/forum/r311931 ... ne-call-Or

7. You can setup SIP Broker on a voice gateway and get access to free calling to over 2,000 VoIP networks for free:
http://www.sipbroker.com
http://www.obitalk.com/forum/index.php?topic=526.0

8. Want to be able to dial into your ATA from a cellphone and have the ATA call you back (great if you have free incoming minute plans) and get access to all of the same services you have on your ATA? You can do that with an Obihai ATA's auto attendant feature: https://www.obitalk.com/forum/index.php?topic=66.0
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.
[OP]
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User avatar
Mar 3, 2002
8179 posts
2409 upvotes
How do I set my Obihai ATA's Ethernet port to 100Mbit full duplex (it's not by default)?

http://www.obihai.com/support/troubleshooting/sg/drop

Dial *** 0
Enter option 27 and press #
Press 1 to set a new value
Enter a value of 1
Press 1 to confirm/save
Hang up
Your ATA should reboot at this point.

I'm noticing static. What should I do?
krazykanuck wrote:
Feb 10th, 2016 11:36 pm
but I do notice a lot of white noise/static on the calls I make.
1. First thing I would do is try a different phone.

2. Some users have reported switching the ethernet port to full duplex fixes static noise. Force your Obihai ATA to use 100Mbit Full duplex by doing the following:

Dial *** 0
Enter option 27 and press #
Press 1 to set a new value
Enter a value of 1
Press 1 to confirm/save
Hang up
Your ATA should reboot at this point.

3. Definitely try moving your ATA away from other devices to see if that helps.

4. Some other people mentioned the power supply adapter being an issue here:
https://www.obitalk.com/forum/index.php?topic=3814.0

But that was for the OBi110/100 series, I think.

If the issue is the adapter, you may be able to get Obihai to send you a new one, but I think I would just ask for a replacement from the point of purchase (probably Newegg Canada). Also visit http://www.obitalk.com/forum/index.php?topic=8754.0


If you try dialing *** and still get static, obviously the issue isn't your VoIP provider.
Please do not PM me for technical assistance unless I PM you first. Please post on the forums instead. I help out when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.51x) can be found here.

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