Expired Hot Deals

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

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  • Dec 22nd, 2018 8:58 am
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FWIW, I'm not sure when the OBi200 deal ends. Maybe Cyber Monday
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OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
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martin_nv wrote:
Nov 28th, 2017 10:03 am
Obi200 for $45 on Amazon right now:
https://www.amazon.ca/OBi200-VoIP-Phone ... rds=obi200
Thanks. SugiStar is not an authorized Obihai retailer in Canada. Only Newegg.ca and Acrovoice are. So you'll be taking your chances with Obihai support, such as it is (under warranty, when your device fails, you have to pay to ship it to them in the U.S. regardless of whether the warranty is honoured).
http://www.obihai.com/how-to-get (select International). You would need to be able to get Obihai to warranty products sold by SugiStar.

Acrovoice stopped carrying the OBi20x series ATAs because Newegg.ca kept undercutting Acrovoice's pricing. So that just leaves Newegg.ca in Canada.
Last edited by Guest1284983 on Nov 28th, 2017 10:08 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.
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Apr 24, 2005
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Does Newegg pay for return shipping on the OBI200 during its warranty period?
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penf wrote:
Nov 28th, 2017 10:32 am
Does Newegg pay for return shipping on the OBI200 during its warranty period?
Nope. The issue is even if you're willing to ship the ATA to Obihai in the U.S., Obihai can deny your warranty claim since you didn't buy the product from an authorized retailer (unless you purchase from Newegg.ca). I have no clue whether they would deny the warranty, but that would be the concern.
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.
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Webslinger wrote:
Nov 28th, 2017 10:36 am
Nope. The issue is even if you're willing to ship the ATA to Obihai in the U.S., Obihai can deny your warranty claim since you didn't buy the product from an authorized retailer (unless you purchase from Newegg.ca). I have no clue whether they would deny the warranty, but that would be the concern.
In your opinion, is the warranty worth it at all? Sounds like a massive pain, especially when you include the fact that you're on the hook for shipping the ATA to the US.
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Elessar wrote:
Nov 28th, 2017 11:37 am
In your opinion, is the warranty worth it at all? Sounds like a massive pain, especially when you include the fact that you're on the hook for shipping the ATA to the US.
It depends on what the problem is. For example, Obihai has been known to ship replacement power adapters to some people for free, when they've experienced issues, without the need
to ship the defective one back. But it's a YMMV (your mileage may vary) situation. And if you have to contact them, for some reason, to get Google Voice working on the Obitalk.com portal
then you probably don't want to be blocked from asking for help, even though you can only do so for free for 1 year without paying $10 USD for support afterwards. I am aware of a situation where someone had to get an Obitalk.com account fixed due to an inability to add a device back to the account after it had been deleted. The solution had to come from Obihai support.

I guess the proper solution is for people to ask Obihai whether they will honour the warranty from a product sold by SugiStar before making a purchase (and then make your own decision after they respond). They can be reached at support@obihai.com. Or visit http://www.obihai.com/contact.
Last edited by Guest1284983 on Nov 28th, 2017 11:57 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.
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Webslinger wrote:
Nov 28th, 2017 11:57 am
I guess the proper solution is for people to ask Obihai whether they will honour the warranty from a product sold by SugiStar before making a purchase (and then make your own decision after they respond). They can be reached at support@obihai.com. Or visit http://www.obihai.com/contact.
Based on your response, I pinged their support and got back a resounding 'No'. They only honor Newegg.
Hi,

The answer is no.

Go to newegg.ca. Such as OBi200 device: https://www.newegg.ca/Product/Product.a ... -_-Product


Best Regards,
Obihai Technology
Now to cancel my Amazon order.
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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 10:06 am, edited 13 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]
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Google Voice ends XMPP support in June '18

https://forums.redflagdeals.com/google- ... #p29271035
Last edited by Guest1284983 on Apr 29th, 2018 12:34 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.
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May 9, 2004
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Upstairs
Webslinger wrote:
Jan 21st, 2018 11:54 pm
Anveo Direct (wholesale service) has pretty decent long distance rates: http://www.anveodirect.com/did/prices

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


...
Hi Webslinger. I stumbled onto this thread last week and found this very interesting so I created an account and added Anveo Direct to my Obi200. Everything seemed to work well until I started placing longer calls and noticed the calls were dropping. I looked at the call logs a few days later and noticed the 4 longer calls I made all dropped at exactly 15 minutes and 32 seconds. I googled this issue and found many hits so it seems to be a known issue. Unfortunately, I was unable to find a solution. It does seem like it may be a configuration issue although I'm unsure what configuration changes can be modified as this is set up as a voice gateway using sip service for voip.ms. Any guidance would be greatly appreciated.
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RedBargainz wrote:
Jul 19th, 2018 2:45 pm
Hi I looked at the call logs a few days later and noticed the 4 longer calls I made all dropped at exactly 15 minutes and 32 seconds.
Anveo Direct is sending a re-invite at the 15 minute mark, and then it waits an additional 32 seconds to receive a reply from your ATA. If the proper response is not received, the call is dropped.

32s drop is a significant time for SIP calls. When they drop at 32s, it usually indicates that an invite was sent by the SIP server and was never received or responded to by your ATA. Typically the reason is due to buggy SIP ALG features in routers, but that may not be the cause in your case. Often, the invite isn't received by the ATA due to some router issue (SIP ALG, NAT problem), but it can also be due to the SIP provider sending the invite into the abyss (to a private LAN IP address instead of your Public WAN IP address).


I have successfully made a call over 22 minutes long using Anveo Direct. My calls are not dropping after 15 minutes and 32 seconds.


Do you have both of these enabled for the SP you're using in step i from the original post?

1.
i) 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)


ii) Also ensure that X_DiscoverPublicAddress is enabled.


2) Also check to see whether SIP ALG is disabled in whatever router you're using. If it's not disabled, try disabling it.
https://www.obitalk.com/info/faq/sip-alg/disable-alg
Read the first 4 pages of the PDF guide (the preamble) found at http://forum.fongo.com/viewtopic.php?f= ... 805#p73839.

a) What brand and model router are you using?

3. If those steps don't help

navigate to Voice Services-->SPx (where X is the # from 1 to 4 of the SP you're using).

These settings are for Freephoneline (and they're probably fine for voip.ms as well):

a) X_KeepAliveEnable: Checked
b) X_KeepAliveExpires: 20
c) X_KeepAliveMsgType: notify
("keep-alive" should be fine for voip.ms)

You can also try asking at http://www.dslreports.com/forum/voip to see if anyone has any further ideas.
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.
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Upstairs
Webslinger, as always, thanks for the quick assistance, much appreciated. Hope all is well with you.

"Do you have both of these enabled for the SP you're using in step i from the original post?

1.
i) 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)

ii) Also ensure that X_DiscoverPublicAddress is enabled."


Yes, already enabled

"a) What brand and model router are you using?"

Not using a standalone router. Using the Virgin Valerie modem which is equivalent to Bell's Hub 3000.

"navigate to Voice Services-->SPx (where X is the # from 1 to 4 of the SP you're using).

These settings are for Freephoneline (and they're probably fine for voip.ms as well):

a) X_KeepAliveEnable: Checked
b) X_KeepAliveExpires: 20
c) X_KeepAliveMsgType: notify
("keep-alive" should be fine for voip.ms)"


Have 3a,3b and 3c set. I changed 3c to "keep-alive" and rebooted and was no longer able to connect to a call...no ringing, just silence after dialing... so changed it back to "notify". Not sure why that would have happened.

I did also change ITSP Profile A > sip > RegisterRetryInterval from 3600 to 120 as a possible solution I found on the web. I have not had a chance to test that yet.

Will report back when I do. Thanks again.

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