Obihai OBi200 VoIP Telephone Adapter with Google Voice & SIP - 49.99

The Obi202 is on $64.99 also!

Does Google Voice work in Canada with these?

I have a Freephoneline.ca SIP, call quality is crap but it’s free and reliable (aka never down).

From what I understand it does for calling out, but you need a US number to get a US GV number (to receive calls).

shayne85 wrote: ↑ Does Google Voice work in Canada with these?

newegg-obihai-obi200-ata-49-99-1-50-ehf ... #p28508756

I have a Freephoneline.ca SIP, call quality is crap

So do I, which I've been using for over 8 years, and the call quality is the same or better than my old Bell POTS (plain old telephone service) landline, which produced static whenever it rained, for local calls. Otherwise, I would not be using it, and I use it daily. Voice quality with any VoIP service is dependent upon jitter (variation between each successive ping, which affects how choppy calls sound), pings (latency), audio codec, and the carrier(s) being used to route the call. Freephoneline, VoIP.ms, Anveo Direct, Callcentric, and Hangouts/GV all sound the same (or better, given I had static over a Bell POTS landline whenever it rained) as a landline to me under ideal conditions for local calls when G.711u is being used, and I do use all of these services. The G.711u audio codec is the equivalent of POTS (plain old telephone service). I don't notice any difference between any of these service providers with respect to call quality for local calling, but I tend to know what I'm doing. International calling to specific numbers is another matter (due to the carriers being used to route the call to the destination). I use Anveo Direct Tier 1 Prime routes for international calls.


VoIP.ms uses Fibernetics (Freephoneline and Fongo's parent company) as one of their (many) carriers.

http://forums.redflagdeals.com/question ... #p25468891

FWIW, Pianoguy is probably the biggest VoIP guru on RFD.

Pianoguy wrote:FreePhoneLine and VoIP.ms provide very similar services. The pricing structure is different, and VoIP.ms has more complex inbound call routing options, but the actual VoIP is the same. VoIP.ms even uses Fibernetics (FreePhoneLine and Fongo's parent company) as one of their carriers, so some of the service is literally identical.

shayne85 wrote:but it’s free

FPL is a one time fee $89.95+tax, unless you're using the desktop app, which is completely free. People usually do have worse experiences with the desktop app.

shayne85 wrote:and reliable (aka never down)

No residential SIP service has 100% uptime.


FPL's last outage was Jun 26, 2018 from 10:20 a.m. EST until 11:05 a.m. EST.

Before that there was an outage on Tuesday, January 30, 2018 between 12:30 p.m. EST and about 3 p.m. EST.

"We will be performing an extensive upgrade to our backend voice calling system on March 28 and 29, 2017 from 12:00AM-06:00AM EST. Voice calling may be intermittently disrupted; please do not submit a support ticket during these times." During part of March 28th, 2017 voicemail notifications were down.

They had a very brief outage on the morning of Dec. 23, 2016 that was fixed by 10:45 a.m. EST. Afterwards they had a voicemail notification issue that was fixed on Dec. 30th, 2016. And before Dec, 23rd, they had a service disruption on the morning October 8th, 2015, and the problem was resolved by 8 a.m EST. That's over a year of up time.

Otherwise, they have been up 24/7. Freephoneline server status information can be found at https://status.fongo.com/.







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.

FWIW, Anveo offers special rates for Obitalk.com users: https://www.anveo.com/anveoforobitalk.asp (prices are in USD).

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.


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

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

I'm not a big fan of this site, but for a general QoS description, visit http://www.voipmechanic.com/qos-for-voip.htm (avoid anything it says about G729 codec).

I'm suggesting Toronto below, because FPL's SIP servers are in Ontario. When you test, pick the location that is closest to your VoIP service provider's server location.

sfrancis wrote: ↑Have been using freephoneline with obihai 202 behind Asus RT N66U router. Often time people complain that my voice very choppy on their end, yet I seem to hear them fine. Is there setting that I should tweak on ATA set up or on router? Thanks very much

That's usually related to upload jitter/packet loss.

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

Probably in the traffic manager area, use a drop down box to select "User Defined Rules". Then create a rule giving traffic highest priority to your ATA's MAC address.

2) Another possibility is you're dealing with congestion during prime time (8p.m. to midnight, especially on Sundays). That's an ISP issue (possibly oversold its service in your area).

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

Pick the test location that's closest to where your VoIP server is situated.
A MOS score below 4.0 is bad news. It means call quality will not be good.
The advanced (+) tab provides interesting info.

You should also try the winmtr test I mention over here around 8 p.m. to FPL's servers:
http://forums.redflagdeals.com/newegg-o ... #p27515963

Anything over 200ms is unacceptable.

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

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

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

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

Navon01 wrote: ↑The Internet is 100down/10up.

Down doesn't matter. What people hear from you is upload.

No other devices are being used at time of bad voice quality.

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

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

Try running http://vac.visualware.com/index.html at 8p.m. (or between 8 p.m. and 11 p.m.)

Select a test server location that's located close to the SIP server you're using.
A MOS score below 4.0 is bad news. It means call quality will not be good.


You should also try the winmtr test (or if you're on a mac, maybe this helps: https://www.reddit.com/r/TagPro/comment ... tr_on_mac/) I mention over here around 8 p.m. to FPL's servers:
http://forums.redflagdeals.com/newegg-o ... #p27515963

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

For VoIP SIP services, you want

1) a router that does not have a full cone NAT,

Visit https://www.think-like-a-computer.com/2 ... es-of-nat/.
Mango from the Obitalk.com forums writes,
“Use a restricted cone NAT router, and do not use port forwarding or DMZ. Restricted cone NAT will only permit
inbound traffic from the service provider you're registered to. If you have a full cone NAT router, it will allow traffic
from any source. This is probably not what you intend.
If you have a Windows computer, you can test your router using the utility here:
http://www.dslreports.com/forum/remark,22292023. To run it, use stun stun.ekiga.net from a command prompt.”
Essentially, you download the stun-test.zip file; extract the stun.exe file from within the zip file to an easily
accessible location; use an elevated command prompt (visit
http://www.thewindowsclub.com/how-to-ru ... inistrator); change directory (cd) to the
directory or location where you extracted stun.exe (visit
http://www.digitalcitizen.life/command- ... c-commands); and type “stun stun.ekiga.net” without
the quotation marks followed by the enter/return button on your keyboard.
Asus routers, at the time of this writing, produce port restricted cone NAT routers, for example and are fine,
provided you’re using one with Asuswrt-Merlin, third party firmware installed: https://asuswrt.lostrealm.ca/about.

2) a router that lets you disable SIP ALG if it's buggy,

To understand why SIP ALG often causes horrible problems, please visit
http://www.voip-info.org/wiki/view/Routers+SIP+ALG (scroll down to the section on SIP ALG problems).

If you're dealing with a modem/router combo issued by an ISP or a router with SIP ALG forced on, you may have
to use voip4.freephoneline.ca:6060 for the Proxy Server. The purpose of voip4.freephoneline.ca:6060 is to circumvent
faulty SIP ALG features in routers.

3) a router that allows you to set QoS or assign highest priority to your ATA or IP Phone over all other devices on your LAN (local area network),

For a very general description of what QoS can do for you, visit https://www.voipmechanic.com/qos-for-voip.htm.
The basic idea is if you're torrenting or have a bunch of other computers, smartphones, tablets, etc. downloading and uploading (hogging all your available bandwidth), you don't want
your ATA not to have access to enough bandwidth to make or receive calls properly. So QoS or a Bandwidth Monitor feature (which is just another form of QoS) is a really good idea for VoIP users.

I often get an occasional relative complaining to me, "Hey my calls sound choppy." And then when I go visit, some kids are playing MMOs on a computer, while another person is downloading a huge file,
and another person is backing up files to a cloud service all at the same time someone else is trying to talk on the phone. All those devices, without QoS enabled, are fighting over available bandwidth along with the ATA.

and 4) A router that lets you adjust both Unreplied and Assured UDP timeouts.

Thanks to Mango, many of us now understand that in order for ATAs to remain registered and working properly with a VoIP SIP provider like Freephoneline, in particular after power failures, the following conditions must be met:

UDP Unreplied Timeout (in your router) < NAT Keep-alive Interval (in your ATA; for Obihai ATAs this is X_KeepAliveExpires) < UDP Assured Timeout (in your router) < SIP Registration Failure Retry Wait Time (or RegisterRetryInterval in Obihai ATAs)

“<“ means less than.

When a modem leases a new IP address, a problem can arise where prior associations using the old IP address are maintained in the router. When the ATA attempts to communicate using the old IP address, the response is unreplied, and then if the UDP Unreplied timeout is greater than the Keep Alive Interval (and UDP Unreplied timeout is often set to 30 by default in consumer routers) a problem arises where the corrupted connection persists. If UDP Unreplied timeout is, for example, 10, and the NAT Keep Alive Interval is 20, then the corrupted connection will timeout or close. A new connection will be created, and everything will work fine.

Another problem can occur when the Keep-Alive interval is greater than UDP Assured Timeout (often 180 by default in consumer routers): the NAT hole will close due to the ATA not communicating frequently enough with the SIP server. In turn, incoming calls may, intermittently, not reach the ATA. Again, X_Keepalives expires is supposed to be 20 with FPL.

(the above settings are making reference to those in Obihai ATAs)

Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers may be difficult, if not impossible. Asuswrt-Merlin (I would avoid any model below an RT-AC68U), third party firmware for Asus routers, does offer easy access to these two settings, which are found under General–>Tools-->Other settings. My understanding is that third party Tomato firmware has these two settings as well. So if your router supports Tomato firmware, that may be another option. Note that I will not be held accountable any damage resulting from failed firmware updates.


Apparently, Mikrotik routers also allow users to change both Assured and Unreplied UDP timeout settings as well:
https://forums.redflagdeals.com/recomme ... #p28056619 (I've never used them and can't advise buying them or answering questions about them.)

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.



ISPs do not issue customers routers that can do all four things I just listed. Typically it's far better to have your own router with strong QoS functions and a restricted cone NAT firewall,
disable whatever SIP ALG feature is enabled in the router, and stick whatever modem/router combo your ISP gives you into bridge mode. For Bell Hubs, visit please-sticky-how-bypass-bell-hub-use-y ... r-1993629/. For Rogers Hitron, visit https://www.rogers.com/customer/support ... ridgemodem (CGN3 instructions also apply to CODA-4582).

Waiting for this deal for a long time. My OBI100 was out of the service about 4 months. Thanks!

shayne85 wrote: ↑ Does Google Voice work in Canada with these?

I have a Freephoneline.ca SIP, call quality is crap but it’s free and reliable (aka never down).

I'm curius about the quality issues you have ? I can't say I'm using my freephoneline that much, however I'm not experiencing quality issues. I had somes at first when I was still on a basic DSL connection but after that, never had an issue and I have the service for ~ 9 years now

NewLogik wrote: ↑ I'm curius about the quality issues you have ? I can't say I'm using my freephoneline that much, however I'm not experiencing quality issues.

I use it daily, and I have no issues for local calling or faxing (FPL, unofficially, works with T.38 fax protocol). All other things being equal, G.711u=G.711u=G7.11u=POTS (plain old telephone service). There's absolutely no way I'd be using it if sounded worse than my old Bell landline, and I use a lot of different VoIP services including Callcentric, VoIP.ms, and Anveo Direct. I have no problems (normally . . . I did have to get VoIP.ms change carrier routing to a couple of locations) with any of them.
But there are tons of people who don't realize their ISP may be at fault and aren't running QoS on their routers for their ATAs and IP Phones. Or they’re located far away from FPL’s servers in southern Ontario and their ISP uses a horrible route to reach them.

A friend recently was complaining to me that his outbound audio using FPL was cutting out, randomly, when he spoke. Sometimes it worked fine for several minutes--and then outbound would cut out--and eventually return (and then randomly drop out again). The people he was calling were complaining. He would have thought that FPL sucks, but he knows from using FPL perfectly fine over at my place that he has a problem. I go over there, log into his modem, and see that the signal going to his modem is poor. It turns out that the signal was too weak. Rogers had to fix the problem at the head end. Download wasn’t noticeably affected, and upload speed tests looked fine. Tracerts and pathpings (over 100) weren't always showing anything odd. But, very occasionally, ping spikes would occur while sending data. This was especially noticeable during online gaming (MMOs). After Rogers fixed the signal issue, the problem disappeared. And that took about a week to resolve. I can only imagine how annoying trying to resolve that issue with an indie ISP would be (customer has to call the ISP; ISP has to send a ticket to the incumbent ISP, Rogers; Rogers needs to acknowledge the issue and not close the ticket). Some ISPs also oversell capacity and/or throttle during prime time. With cable internet, sometimes local node congestion occurs, in particular during prime time, depending on the location. DSL users can also suffer from congestion issues too. If you don't have solid internet service, VoIP is going to be horrible.

Is the OBi202 @ $65 with Free shipping the best deal on ATAs to buy right now?

I want:

- Reliable;
- Simple (but extra features are always nice);

How likely is it that we are going to see a lower price on this in the next few months?

In other words, is it time to pull the trigger on this to replace an 8 year old Linksys PAP2T ATA before it fails?

Are there any reason NOT to buy the 202 but the 200 instead?

TIA.

Temporel wrote: ↑

How likely is it that we are going to see a lower price on this in the next few months?

I’d say almost 0% based on past history. Next round of sales might be during Boxing week, and I doubt the pricing will be better.

I've stopped posting Obihai ATA deals in the Hot deals forums for this reason:
http://forums.redflagdeals.com/google-v ... #p29320252
http://forums.redflagdeals.com/google-v ... #p29356215
Basically, Obihai OBi2XX ATAs are still the most powerful ATAs intended for residential usage, but their support policies can offend people.

If you want to learn more about the difference between the two ATAs, visit newegg-obihai-obi202-ata-69-99-1-50-ehf ... x-2146968/ and newegg-obihai-obi200-ata-49-99-1-50-ehf ... x-2145415/. In particular, read through the first two pages on each thread.

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).

Has anyone used an Obihai, with any service, for their home alarm system?

More general information can be found from reading through the first few pages of these threads: newegg-obihai-obi202-ata-69-99-1-50-ehf ... x-2146968/ and newegg-obihai-obi200-ata-49-99-1-50-ehf ... x-2145415/.

ceredon wrote: ↑ Has anyone used an Obihai, with any service, for their home alarm system?

From newegg-obihai-obi200-ata-49-99-1-50-ehf ... #p28508726

Can I use VoIP with an alarm monitoring service?



http://forums.redflagdeals.com/free-pho ... #p21992603

fastlayne wrote: ↑I use FPL and The Monitoring Center with an old DSC PC3000. I have two FPL numbers - one for the house phone system and one for the alarm.

I connected the DSC directly to a dedicated VOIP ATA. I did not want the DSC seizing my phone line and for $40 the extra ATA isolates any problems. If I needed to tweak the DSC ATA, I did not have to change my house phone ATA.

I had some problems initially with the useless Cisco SPA112, but have not had any problems with the Obi100.

(Disclaimer: I assume all risks in using VOIP for my alarm system. Thanks to everyone that is worried about my well being.)

http://forums.redflagdeals.com/need-lan ... #p23753017

fastlayne wrote: ↑My Freephoneline, Obi100 ATA, and DSC PC3000 (?) all play nice together.

It is very easy, at least for DSC, to use a RJ45 straight coupler to connect directly to the ATA with a short length of RJ11 cable.

This creates a dedicated configuration without having to play with any house wiring or the RJ31X and gives the alarm system its own "phone line".

http://forums.redflagdeals.com/carrytel ... #p27166801

kanata2004 wrote:I use freephone line + Grandstream HT502 and it works great with my alarm company (ADT). Also I tried the Fido home phone (gsm based) and it works fine too. I went with Freephoneline way since no monthly cost. I tried FPL + Obi200 device and it works just fine too. Search RFD if you want know more obi200 and Freephoneline . There are thousands of posts.

You can make the alarm system work first and switch internet after.

Freephoneline users may find this thread useful: http://forum.fongo.com/viewtopic.php?f=8&t=19236.




This post might be a good starting point: http://forums.redflagdeals.com/voip-ms- ... #p16112120


Anyway, clearly, it's possible:


http://forums.redflagdeals.com/voip-ms- ... st16144590

mintchoco is using voip.ms with The Monitoring Centre with an Obihai ATA.

http://forums.redflagdeals.com/voip-ms- ... #p22798293

Willyburan is using Callcentric (another popular VoIP service provider) with an OBi110

http://forums.redflagdeals.com/voip-hom ... #p16383484

c-ditty is using voip.ms with a Linksys PAP2T






But using VoIP may be less reliable than using a landline for an alarm monitoring service, except you should take note of what Pianoguy mentions below.

Setting up a VoIP service with alarm systems is not especially easy for everyone. So, if you're going to attempt this yourself, don't except guaranteed success.




http://forums.redflagdeals.com/free-pho ... #p23087011

Pianoguy wrote: The service provider is not really relevant because everyone uses the G.711u codec. Setting up your equipment correctly will make the difference.

AcroVoice offers support for alarm systems, and they will preconfigure your device so that it is ready to go as soon as you receive it. If you want to DIY your VoIP setup, the OBi200 and OBi202 ATAs allow you to configure a fixed jitter buffer, which is crucial for reliable data transmission. Look in the admin guide for $NOJI1 and also $NOEC1.

Acrovoice's website can be found here: https://www.acrovoice.ca/content/residential_service

http://forums.redflagdeals.com/grand-al ... #p17534879

Pianoguy wrote:I would make the argument that an alarm system connected to a POTS line is not reliable. If your demarc is mounted outdoors, an intruder could simply disconnect the line.

Instead, why not buy a system that connects to the monitoring station via cell, and perhaps also via IP so you can monitor it from your phone?

If you have a smartphone, this device will allow you to monitor your alarm with it. No fees beyond equipment purchase and installation:
https://www.eyez-on.com/EZMAIN/envisalink3.php

You can receive a text message when your alarm trips, and there are no monthly fees. The downside is no police dispatch, but (at least in Vancouver) it's slow enough that there's not a great deal of value there.

It should also be noted that many alarm systems have batteries. So if there's a really long power outage, your alarm system may not work either, depending on the system.

ceredon wrote: ↑ Has anyone used an Obihai, with any service, for their home alarm system?

When I first started using an Obi (with FPL) a couple years ago, I tried to get my alarm to work with it. I was actually able to make it work (think I may have routed the outbound call through Google Hangouts, but can’t remember). It did work sometimes, but there were times where it wouldn’t for whatever reason and my alarm company would contact me to let me know it was disconnected. Ultimately, my alarm contract was up and I was able to renew and have them throw in the cellular alarm chip for free so I went with that.

No complaints about the Obi / FPL / Google Hangouts. Works just as well as a POTS phone as others have mentioned and I am quite happy with no recurring bill for my home phone service.

Temporel wrote: ↑ Is the OBi202 @ $65 with Free shipping the best deal on ATAs to buy right now?

I want:

- Reliable;
- Simple (but extra features are always nice);

How likely is it that we are going to see a lower price on this in the next few months?

In other words, is it time to pull the trigger on this to replace an 8 year old Linksys PAP2T ATA before it fails?

Are there any reason NOT to buy the 202 but the 200 instead?

TIA.

I have also been using my Linksys PAP2T with voip.ms for like 6 years.
What makes you think your PAP2T will fail? Mine has been problem free all those years.

How do you use this with Freephoneline and Google voice-US number with one tel port?

Have Linksys PAP2T ATA with Freephoneline for 7 years. Was using the second port of ATA for Google voice with Simonics workaround for some time. Now they have stopped supporting Google voice.

new936 wrote: ↑ How do you use this with Freephoneline and Google voice-US number with one tel port?

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

I suggest reading the first two pages on this thread for answers to general questions: newegg-obihai-obi200-ata-49-99-1-50-ehf ... x-2145415/.

Simonics workaround for some time . . . Now they have stopped supporting Google voice.

This may be of interest: https://www.dslreports.com/forum/r32134 ... -Pbxes-com. I don’t use it and won’t be answering questions about it, but if you have questions you can ask there.

AMD wrote: ↑ I have also been using my Linksys PAP2T with voip.ms for like 6 years.
What makes you think your PAP2T will fail? Mine has been problem free all those years.

My PAP2T died a few month ago...

richardjhy wrote: ↑ My PAP2T died a few month ago...

How old was it? Died all of a sudden or was acting up before kicking the bucket?

I'm considering this deal but my Linksys PAP2-NA is still going strong. The PAP2T version can be found on ebay for under $30 as well.