Expired Hot Deals

[Newegg] Obihai OBi202 ATA $69.99 + $1.50 EHF+ free shipping+tax

  • Last Updated:
  • Dec 22nd, 2018 8:58 am
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3264 upvotes
How do I remove the requirement to dial **2 to dial out on SP2?

For people wanting to remove the requirement to dial **2 before the phone number to dial out using the service provider on SP2, for example, you might want to ask X360 for help. This post might be useful to you: http://forums.redflagdeals.com/freephon ... #p20004155

If you just want to press "1" to dial out on SP2, visit http://forums.redflagdeals.com/freephon ... #p24412847

You might also want to work your way through my responses to RedBargainz: no-incoming-calls-google-voice-2019683/3/#p26535001.

Otherwise, ask at https://www.obitalk.com/forum/index.php?board=3.0.

Please don't ask me to configure your dialplan; it's just too much work.


Did you know you can 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 tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3264 upvotes
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:
Mar 30th, 2016 11:28 pm
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:
Apr 7th, 2016 11:17 pm
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.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3264 upvotes
I'm having trouble with not receiving incoming calls to Google Voice. What should I do?

First, you need a Google Voice phone number. I won't be assisting people in getting one.

Make sure your Obihai ATA firmware is updated.

Then visit http://www.obitalk.com/forum/index.php? ... 0#msg56460 and read that post fully.


Also visit http://www.obihai.com/support/troubleshooting/sg/inc
Click the link, select the "I am using Google Voice" option, and look at the pic.



"Check" Google Chat.


1. Sign in to your Google Voice account
2. Go to "Settings" located on the top right (cog wheel)
3. On the Phone tab, you will see a "Delete" option under Google Chat
4. Click "Delete" Google Chat
5. Open a new browser and sign in to www.gmail.com using your Gmail account. You will see Gmail Call Phone (located on the bottom left corner).
6. Make a call out from your Gmail Call Phone
7. Once you have done so, Google Chat should appear again in https://www.google.com/voice#phones
8. Make sure to check the checkbox for Google Chat


If you are still not able to receive incoming calls, check if your email underneath Google Chat is an @gmail.com account and/or delete and re-add Google Chat.

Image
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3264 upvotes
Why am I not receiving Caller ID Name information on all incoming calls?


CNAM lookups are dependent upon your VoIP service provider--and not the Obihai ATA.

Freephoneline doesn't do CNAM lookups (I.E., Freephoneline doesn't lookup the name of the caller, using the incoming phone number, in an external database). Unless CNAM is being sent by the call provider, which often is not the case with cellular calls, only the phone numbers are going to appear. CNAM lookups cost money, and for a moderately priced one time fee, FPL isn't going to do CNAM lookups.

http://forum.fongo.com/viewtopic.php?f= ... 68&p=25678 (a lot of replies in that thread are incorrect; CNAM will show up if it's sent by the provider)
"Just a reminder, as has been provided in other threads, when the call display feature is enabled, it will only show the information that is being sent by the provider. If that provider only sends a number, we can only display a number. This is especially common for cell phone providers and a number of digital service providers."--Fongo_mike"

So, with an incoming call to your FPL #, an incoming phone number will appear on your callerid, but a name only shows if that info is also sent by the provider. Other VoIP services typically charge a fee per call to do a lookup in a CNAM database (phone number is queried for a name in a database). FPL doesn't offer that option. If CNAM info is passed on by the provider, FPL will display CNAM.



With VoIP.ms, CNAM is only guaranteed with Premium routes.

Google Voice doesn't support outbound CNAM. If something ever displays in the name field from a GV call, the CNAM info was populated by the recipient's service provider, which would have likely done a CNAM lookup from a database. You cannot simply add your name either for your name to show up when using GV. You may be able to get your name listed in a CNAM database (often by paying), but there's no guarantee the provider being used by the person you're calling is using that database. Refer to http://classroom.synonym.com/comes-up-c ... 13680.html
GV also doesn't support inbound CNAM.


Freephoneline, VoIP.ms, Anveo, Callcentric, etc. support outbound and inbound CNAM.
But FPL will not do CNAM lookups (which cost money).
Last edited by Guest1284983 on Nov 23rd, 2018 9:10 am, edited 1 time in total.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3264 upvotes
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 instead of paying $1.50 USD/month with Callcentric, Anveo, 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 Guest1284983 on Dec 18th, 2018 10:44 am, edited 1 time in total.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
Deal Addict
User avatar
Sep 2, 2010
1854 posts
984 upvotes
montreal
deal link is wrong (it was from your session i believe.

Correct link is:
obi202

Edit: big fingers on cell phone... And need more coffee.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3264 upvotes
superbigjay wrote:
Nov 24th, 2017 7:44 am
deal link is wrong (it was from your session i believe.
Fixed. Thank you
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3264 upvotes
Looks as though this deal has been extended to Cyber Monday.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3264 upvotes
Sale has been extended again until Thursday.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3264 upvotes
Anveo Direct (wholesale service) has pretty decent long distance rates: http://www.anveodirect.com/did/prices


* Note that the following works using a router running Asuswrt-Merlin with SIP Passthrough set to "Enabled + NAT Helper".
It may not work with other routers, depending on the way SIP ALG operates in them. When SIP Passthrough in Asuswrt-Merlin is set to disabled, Anveo Direct's SIP trace shows that the contact header contains the Private LAN IP address of the Obihai ATA and not the WAN IP. In this situation, Anveo Direct eventually sends a re-invite at the 15 minute mark and because no ACK response is received, at 15 minutes and 32 seconds, the call will drop. This problem also occurs when SIP Passthrough is set to "Enabled". So, this is a rare situation when SIP ALG actually helps, provided it can replace LAN IPs with WAN IPs in the contact header. When SIP Passthrough is set to "Enabled + NAT Helper" using Asuswrt-Merlin, the contact field contains the WAN IP address, and invites are responded to and received.

** I've now added instructions for those without Asuswrt-Merlin.

How Do I Setup Anveo Direct as a Voice Gateway on an OBi2xx series ATA? (for outgoing calls only)


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

In this example, A SIP service (VoIP.ms, Anveo, FPL, etc.) is setup on SP2. This is important to keep track of. For a Voice Gateway to work, another SIP Service/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.

First, you create an Anveo Direct account. I think they give you $0.60 USD to play around with. You will then need to configure an Outbound Trunk in your Anveo Direct account (website).
Title can be whatever you want.
Dialing Prefix is whatever 6 digits you want starting with 0.
Authorized IP is your Public WAN IP address.
Concurrent Call limit is probably 2 (depending on the SP you're using and another Obihai setting)
The rest of the stuff you should be able to figure out for yourself (choose what you want). Sorry, I'm not going to start answering questions about their routes and carriers.
After you're done, click save.

Whenever your Public WAN IP changes, you need to update it in Anveo Direct's user portal.



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.)


A. Navigate to Service Providers-->ITSP Profile X (whatever SP you're going to be using for outbound SIP URI)-->SIP-->X_UsePublicAddressInVia
enable X_UsePublicAddressInVia (click save)

Also ensure that X_DiscoverPublicAddress is enabled.

B. For those without a router running Asuswrt-Merlin with SIP Passthrough set to "Enabled + NAT Helper" (that is, SIP ALG is off)
i. Navigate to Service Providers--> ITSP Profile X (where X is whatever you're using)-->General
ii. Enable STUNEnable
iii. for STUNServer, try stun.callwithus.com
(you can google a list of public stun servers to try)
(save settings)



1.
a. Navigate to Voice Services-->Gateway and Trunk Groups
b. Select an unused Voice Gateway
c. Enable check
d. Name Anveo Direct
e. AccessNumber is SP2(sbc.anveo.com)

If you don't have Asuswrt-Merlin, use SP2(sbc.anveo.com;op=sn) for AccessNumber. Remember that stun needs to be enabled in the ATA.

I tried enabling STUN in the ATA with SIP ALG disabled, and it appears to work for two-way audio with Anveo Direct.
In the event the STUN server drops, you'll encounter problems with both Freephoneline and Anveo Direct.
Unfortunately, I don't believe Obihai ever implemented this suggestion: https://www.obitalk.com/forum/index.php?topic=440.0.


d. DigitMap (XX.)

XX. stands for any phone number you can punch in. If you know what you're doing, change that to what you want. Otherwise, leave it alone.

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.

If you have SIP service setup on SP1, then use SP1(sbc.anveo.com)

e. AuthUserID enter what you want for an outbound CID number. ex. 15191234567

f. You don't need to enter anything for Authpassword.

(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|)

You can change *2 to whatever you want, but it can't have been used before by you for something else.


((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 basically a 3 second delay while the ATA waits for you to punch in the phone number.

You're dialing *2 to dial out on Anveo Direct in this example.

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

c. You need to add

{(<*2:>(Mvgx)):vgx(>012345*$2)}

*2 is what you dial before the phone number (corresponds with what you created in step 2a). When you dial *2, it's replaced by 012345 in front of the phone number you dial.
012345 is the Anveo Direct prefix that you create (you'll need to change that 6 digit prefix to the prefix you created in your Anveo Direct outbound route).

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},{(<*2:>(Mvg2)):vg2(>012345*$2)},{(<**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 "2". You'll need to change 012345 to the 6 digit prefix you're actually using with Anveo Direct.

d. submit/save/reboot

And now you dial *2 + phone number to dial out on Anveo Direct.




Anyway, that should work provided you don't have some firewall setup where you need to use STUN.

Anveo Direct has an API that might be able to be used to update WAN IP automatically if you're running your own server or another device, but I won't be able to help with that. I'm having trouble enough with trying to figure out a workaround on my own; so if someone has a step by step solution (cron job using a script with Asus Merlin using the Anveo Direct API?) for automatically updating WAN IP in Anveo Direct's user portal, please post it. Thanks.
Last edited by Guest1284983 on Jul 29th, 2018 10:47 am, edited 9 times in total.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3264 upvotes
Google Voice ends XMPP support in June '18

https://forums.redflagdeals.com/google- ... #p29271035
Last edited by Guest1284983 on Apr 29th, 2018 12:35 am, edited 2 times in total.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
Deal Addict
May 23, 2008
1741 posts
253 upvotes
Vaughan, Ontario
Webslinger wrote:
Apr 29th, 2018 12:35 am
Google Voice ends XMPP support in June '18 (Google Voice Gateway and Obihai hacked firmware users potentially affected)

Visit http://www.obitalk.com/forum/index.php?topic=13824.0.

Unless someone can develop a fix/support, people using this hacked firmware will be affected: http://www.obifirmware.com/. That could especially be an issue
for Obihai 1xx ATA users.

Bill Simon's Google Voice Gateway customers may also want to pay attention: http://www.dslreports.com/forum/r31938647-.
https://simonics.com/gw/

I expect this situation will require people, including OBi2xx users, to ensure Obitalk service and Obitalk Provisioning are enabled again. For those that disabled those settings, you may have to visit
https://forums.redflagdeals.com/newegg- ... #p28531310 and enable them (do the reverse) once new firmware is issued.
(Publicly, no one knows, at the time of this post, for sure).
Be sure to backup your existing ATA settings first before enabling Obitalk Provisioning (the settings from Obitalk.com will be transferred to your ATA).
My 202's support expired last December. When I logged in via online portal tonight, I saw a yellow triangle alerting me there was a firmware update available. I thought there was an unannounced one after 3.1.1 5804, so I proceeded with the update. In a few minutes, my 202 rolled back to 3.2.1 5757EX. It was very interesting. Did it mean Obi would extend support on my 202? Why is the official fw for 202 still 3.2.1 5757ex?

I manually updated to 3.1.1 5804 after.
[OP]
Deal Fanatic
User avatar
Mar 3, 2002
9417 posts
3264 upvotes
ma678 wrote:
Apr 29th, 2018 1:00 am
Did it mean Obi would extend support on my 202?
No. I strongly suspect that just means Obihai is willing to push specific firmware that will ensure users continued support of Google Voice. That's it.
That's what happened before.
Why is the official fw for 202 still 3.2.1 5757ex
It's not. That's the firmware version that was getting pushed to people who have a device that's not in warranty or not covered by extended support.
As you've noticed, there's a newer firmware version available. Currently, the latest version is 3.2.2 (Build: 5859EX).

Navigate to System Management--> Auto Provisioning-->Auto Firmware Update
In the FirmwareURL field blank out whatever is there.

Then try updating firmware again.
Last edited by Guest1284983 on Apr 29th, 2018 1:06 am, edited 1 time in total.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.

Top