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[Newegg] Obihai OBi200 - $59.99

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  • Nov 27th, 2015 2:30 pm
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Deal Addict
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Dec 22, 2014
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Mordor
Great! Probably the cheapest in Canada ever?
I think it was 29.99 is US. Wish we had that deal
Deal Fanatic
Oct 18, 2004
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Wat
ShadowMordor wrote: Great! Probably the cheapest in Canada ever?
I think it was 29.99 is US. Wish we had that deal
I think it was $49.99 earlier in the year. But still a good price, to sway the people thinking about Ooma.
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Jul 15, 2003
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Regardless of wether it's the "best" price, this is a GREAT price for an amazing piece of hardware. If you're thinking about voip..... GET IT!

You can think about and decide on which voip provider you'll use later.
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Dec 22, 2014
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Mordor
Kevinck wrote: Regardless of wether it's the "best" price, this is a GREAT price for an amazing piece of hardware. If you're thinking about voip..... GET IT!
+1
Deal Addict
Aug 16, 2003
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I got one from Amazon in the US for $45 (so the same price as this deal) and use it with Google Voice.

Works well and is simple to manage.
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Nov 28, 2013
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Oakville
And well then, there we have exactly what I was looking for the other day. Hrmm.
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Sep 10, 2005
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Thinking of getting this to replace the Linksys PAP2T I've been using with voip.ms and go with FPL. What do you guys think?

I've been satisfied with voip.ms but the fact they charge in USD and the Canadian dollar being where it is, it doesn't seem worth it to keep topping up with them.
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Mar 3, 2002
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Dave98 wrote: Thinking of getting this to replace the Linksys PAP2T I've been using with voip.ms and go with FPL. What do you guys think?
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 used to get very low latency to voip.ms closest sip servers to me as well, but recently they've climbed to 48 ms (still perfectly fine). I'm still not detecting any delay.
http://wiki.voip.ms/article/Choosing_Server
In fact, I'm getting lower pings than 48 ms to Chicago and New York (around 24 ms).


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



I use FPL and voip.ms (and lots of other providers). You're not stuck with a single provider with an Obi200 (I use the Obi202).
You do not have to pay for a DID to make outbound calls with VoIP.ms

OBi200/202 offers support for up to 4 SIP trunks and 8 voice gateways.

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.

Freephoneline is another account. You get a Canadian phone number. You can call to most major Canadian cities for free.
$60 setup fee+tax. No ongoing fees for as long as you use FPL. But each e911 call is $35+tax (however, consider that
all other providers charge a monthly fee for this. If you don't more than one 911 call every two/three years, you're laughing).

Someone might be using VoiP.ms on a voice gateway for outgoing calls only.
Someone might be using a VoIP.ms phone number on a SIP account.

Some people use free N.Y. phone numbers from Callcentric. That's another SIP account.

Some people might be using Anveo.

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
I would prefer people not PMing me or asking me how to set that up though. Sorry.

Someone might also want a free U.K. phone number: https://www.telng.com/ukddi.html
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Mar 3, 2002
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Image (USD pricing)

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

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

- 1 phone port

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

- 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)

- USB (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

- 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)--and access to OBiPlus.

(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
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T.38 fax protocol works with Freephoneline and an Obi200/202 ATA.

VoIP.ms and Anveo (retail) only support T.38 fax protocol on the backend via their respective online fax web portals.


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

1) Use the latest firmware release (currently 3-0-1-4822 at the time of this post): http://fw.obihai.com/OBi202-3-0-1-4822.fw
http://www.obitalk.com/forum/index.php?topic=8982.0

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, and

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

6)T38Enable should be checked

(4, 5, and 6 should be default settings)

7) T38ECM is checked for me (and seems to work). This is not a default setting. If you have problems with this checked, uncheck it.


(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
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I'm going to copy some of my older posts here, to help curtail PMs.


Having problems with SIP Scanners? Is your phone ringing constantly with caller ids that appear as 1001, 999, etc. Bots/hackers/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.
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.


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?

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.

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.
merged-freephoneline-ca-free-local-soft ... st21807239
merged-freephoneline-ca-free-local-soft ... st21660123
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: merged-freephoneline-ca-free-local-soft ... st21807239 Edit: It appears that OBiPlus Basic is no longer being offered for new customers.


Also, you 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.

Good luck!
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I'm going to copy some of my older posts here, to help curtail PMs.
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

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)
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I'm going to copy some of my older posts here, to help curtail PMs.
T-Bone wrote: 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).

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

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.

(submit/save/reboot)

4. Navigate to Service Providers --> ITSP Profile (FPL) --> SIP
5. For Proxy server, enter freephoneline.ca
6. Registrar server should be blank
7. enable X_ProxyServerRedundancy
(submit/save/reboot)
Jeff146 wrote: 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)
Last edited by Guest1284983 on Oct 5th, 2015 10:15 am, edited 1 time in total.
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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 broswer

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.
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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).

The lastest firmware, at the time of this post is 3-0-1-4822: http://fw.obihai.com/OBi202-3-0-1-4822.fw
(and yes, that's the same firmware used for the OBi200)

Manual update instructions are located here if you don't want to use the web portal: http://www.obitalk.com/forum/index.php?topic=9.0
Deal Addict
Dec 10, 2005
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I used to support OBI. Then they start to charge $10 for firmware upgrade so that older models can continue to use google voice. Although it is still possible to manual flash it if you are willing to take the trouble to do research etc, I would avoid this company.

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