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

qweasdzxc63 wrote: ↑ I have Obihai110 and not long work with GV. Does OBi200 support GV and for how long?

The 200 works with GV (I have 2 GV lines on an Obi200). No one knows for sure how long GV will be supported on the Obi200. If you need GV, you may as well get it at this price.

qweasdzxc63 wrote: ↑ I have Obihai110 and not long work with GV.

You can use https://www.dslreports.com/forum/r32134 ... -Pbxes-com. (I will not be responding to questions about it.) Also, http://www.dslreports.com/forum/r32191614- may be of interest to OBi1xx users (again, ask over there).

Thank Op, I ordered one. Any particular router work best for this Obi? I'm currently looking at $139 ASUS AC1900 Dual-Band Router (RT-AC1900P).

bumbum007 wrote: ↑ Thank Op, I ordered one. Any particular router work best for this Obi? I'm currently looking at $139 ASUS AC1900 Dual-Band Router (RT-AC1900P).

I'm not an expert like Websligner but I bet any decent router will do. What's more important, I think, is a stable Internet connection with low latency and low jitter.

NorthernPaladin wrote: ↑ I own / support about a half dozen Linksys PAP2T's. Over the years I have had a few power supplies go bad which can look like a dead PAP2T (only red lights on or flashing etc) but it was actually a failing power supply. I have found that these PAP2T's are very sensitive to voltage range and anything outside the narrow band they are expecting will cause them to to not work (usually indicated by red light). If you can test with a known / good working PSU (from another PAP2T) you may find that your PAP2T is actually working fine.... Just to note, I have had issues with some psu's from other devices rated the same voltage/amps as the PAP2T unit still not work with the PAP2T....

Thanks. I've tried with other good PSUs. Also I measured PSU voltages with a decent multimeter. All PSUs are in very good condition.

Temporel wrote: ↑ I'm not an expert like Websligner but I bet any decent router will do.

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:
recommendations-new-router-2115672/2/#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 http://forums.redflagdeals.com/please-s ... r-1993629/. For Rogers Hitron, visit https://www.rogers.com/customer/support ... ridgemodem (CGN3 instructions also apply to CODA-4582).

bumbum007 wrote: ↑ I'm currently looking at $139 ASUS AC1900 Dual-Band Router (RT-AC1900P).

Once you install Asuswrt-Merlin, that router satisfies the four criteria listed here: newegg-obihai-obi200-voip-telephone-ada ... #p30145487. So that's a reasonable choice for use with SIP services.

Also, if you have a relatively static WAN IP address, you can use Anveo Direct for international calls without having to specify a STUN server when using that router: newegg-obihai-obi200-ata-49-99-1-50-ehf ... #p28835257. Anveo Direct has pretty aggressive international calling rates.

Lol! I wanted to save you from having to post a reply, but i think i will just get out of the way Master Webslinger.

After my PAP2T dead, I switched back to a very old router-ATA adapter combo Linksys WRTP54G on which LAN ports only support 10mbps duplex now, but ATA adapter is still working fine.

richardjhy wrote: ↑ After my PAP2T dead, I switched back to a very old router-ATA adapter combo Linksys WRTP54G on which LAN ports only support 10mbps duplex now, but ATA adapter is still working fine.

That's exactly my set up also. WRT54G + PAP2T
Been working flawlessly for years.

You mean WRT54G, right?

Temporel wrote: ↑ That's exactly my set up also. WRT54G + PAP2T
Been working flawlessly for years.

You mean WRT54G, right?

No. WRTP54G

Temporel wrote: ↑ Lol! I wanted to save you from having to post a reply, but i think i will just get out of the way Master Webslinger.

Thank you. FWIW, I don't claim to be an expert at much of anything, including Obihai ATAs or SIP services.

Webslinger wrote: ↑ Anveo Direct has pretty aggressive international calling rates.

...but always check VOIP.MS rates for the same destination; I found out some times there is a half price difference... Also, some Prime rates on VOIP.MS are sometimes better than the Standard rates so check out both.

Webslinger wrote: ↑ Thank you. FWIW, I don't claim to be an expert at much of anything, including Obihai ATAs or SIP services.

Are you kidding? They say if you know more than 95% of the people on a subject, you are an expert.
I say (without brown-nosing) that you know more than 99.9% of the people on that subject. What does that make you?
I would probably have to talk with >1,000 persons before I find someone who would know more than you on this subject.
I'm writing a "Thank You!" in the name of many people over here I'm sure for your positive contribution to this community.

max011 wrote: ↑ ...but always check VOIP.MS rates for the same destination; I found out some times there is a half price difference... Also, some Prime rates on VOIP.MS are sometimes better than the Standard rates so check out both.

Can you give me an example of where VoiP.ms’ international rates are better than Anveo Direct’s? Because I can’t think of any off the top of my head:
http://www.anveodirect.com/prices/outbound

Webslinger wrote: ↑ Can you give me an example of where VoiP.ms’ international rates are better than Anveo Direct’s? Because I can’t
http://www.anveodirect.com/prices/outbound

Off the top of my head, Serbia [381xx] cuz I had to call someone there the other day...

max011 wrote: ↑ Off the top of my head, Serbia [381xx] cuz I had to call someone there the other day...

Wow. Thanks, I've never called Serbia. Well, I still have funds on my VoIP.ms account, but many of the international destinations I call are cheaper using Anveo Direct.

Webslinger wrote: ↑ Wow. Thanks, I've never called Serbia. Well, I still have funds on my VoIP.ms account.

Well then, I can give you some numbers over there if you want!! LOL

I I have used magic jack for the last 2 years no issues...I think MJ is better alternative

maliken7576 wrote: ↑ I I have used magic jack for the last 2 years no issues...I think MJ is better alternative

Before you get flamed, I'd just say "there might be some uses for MJ for some people" but it DOES NOT even compare to what we are talking about here. If it works for you - great!