[Newegg] Obihai OBi202 VoIP Phone Adapter $64.99

Before misinformation is posted again . . .

There's two ways to update firmware (for free even if the ATA is out of warranty)--without using the Obitalk.com web portal:

1. Dialing ***6 (and then pressing "1" if an update is available)

http://www.obihai.com/docs/OBiProvisioningGuide.pdf (page 15)

This is a faster method than using the Obitalk web portal since you don't have to log into anything.

2. Manually updating firmware via the device:
http://www.obihai.com/docs/OBiDeviceAdminGuide.pdf (page 43)

Also found after visiting http://www.obihai.com/docs-downloads and clicking on "OBi Device Firmware"

In other words, the other firmware update methods are documented both online and in the device manuals.

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 latest firmware at the time of this post is 3.1.0 (Build: 5264): http://fw.obihai.com/OBi202-3-1-0-5264.fw (also works with OBi200). Note that if you have OBiPLUS firmware installed already and if your subscription is expired, 3.1.0.5264 will completely kill OBiPLUS forever.

Avoid 3-0-1-4972: http://www.obitalk.com/forum/index.php?topic=10520.0

--


This comes directly from Obihai Support:

a) Will Obihai continue to allow its customers to update firmware manually without using Obitalk.com if their devices are out of warranty and without paying $10 annually? Yes/No

Answer: YES


b) Will customers who do not pay $10 annually still be able to dial ***6 to update firmware without using the Obitalk.com web portal?

Answer: yes

c) Is Obihai planning to completely block customers who do not pay $10 annually from updating device firmware manually and also block manual firmware file downloads?

Answer: We don't do any blockings.


--

Obihai is charging after one year to use their web portal to upgrade firmware (via the web portal)--or for support. That fee has nothing to do with activating or configuring Google Voice, which can still be done, for free, even if the device is out of warranty after one year. And firmware can still be updated manually after one year as well, for free, as well.

Read: http://www.obitalk.com/forum/index.php? ... 2#msg66682

"Regarding the new extended support fee, here are some facts:

The optional fee would cover 1:1 customer support from Obihai for another year, along with firmware updates downloaded and applied via clicking the yellow triangle on the portal site.
Obihai is not locking anyone out from configuring Google Voice via the portal.

regardless of warranty status, the latest firmware is available for anyone to download and install manually, at no charge, at this post: http://www.obitalk.com/forum/index.php?topic=9.0
Devices which have been manually-updated to the current firmware level can be added to the portal and configured as usual, regardless of warranty status.
The new extended support fee is no different than most manufacturers' policies regarding providing customers with 1:1 support post-warranty, and, as long as you don't need that kind of help, there is no "ransom" or mandatory charge to continue using OBi devices.
See this discussion for instructions: http://www.obitalk.com/forum/index.php?topic=10065.0


Note: if your OBi is currently within its original one-year hardware warranty period, the portal will offer you an optional one-year extended hardware warranty, plus extended customer support (for $20 or $30 US, depending on the device). If your OBi is currently past its original one-year hardware warranty, the portal will offer you optional extended 1:1 direct customer support for $10 instead. The former option would include service for failed hardware (e.g. device replacement), whereas the latter would include just technical support and firmware updating via the portal. The key word here is "optional".

-- SteveInWA (Obihai beta tester)

(USD pricing)

OBi202

- 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)

- has 2 phone ports

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

- offers 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)

- offers USB port (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 (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)--and access to OBiPlus (Note that OBiPlus is no longer available to new customers; it is grandfathered to existing OBiPlus customers though).

(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?”

The OBi202 (black box) in this video is the ATA.

What really sets Obihai ATAs apart, in addition to being able to use Google Voice, is their powerful call routing features.

You can have 4 different phone numbers (or more) from 4 different service providers. But if you just want to make outgoing calls without wanting to pay for a DID/phone number from a specific provider (VoIP.ms, for example), you can also make use of up to 8 voice gateways. So you could be using 12 different service providers at the same time, all on the same phone, if you want.

You could be using Freephoneline, Google Voice, Voip.MS, Anveo, Callcentric, etc. all on the same phone line.

I use FPL and voip.ms (and lots of other providers). You're not stuck with a single provider with an Obihai ATA.
You do not have to pay for a DID (phone number) 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.
Note that obtaining a Google Voice number requires a U.S. IP address and a legitimate U.S. phone number (typically not VoIP) that hasn't been used to activate Google Voice before. I will not be helping people obtain Google Voice phone numbers, sorry.

Freephoneline is another account. You get a Canadian phone number. You can call to most major Canadian cities for free.
$80 setup fee+tax. No ongoing fees for as long as you use FPL. Each 911 call costs $35 with FPL. If you dial 911 less than twice a year (or less than every 3 years with Anveo's $0.80 USD/monthly fee) vs. paying $1.50 USD/month with Callcentric 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.)


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. Anveo Obihai special pricing can be found here (in USD): http://www.anveo.com/anveoforobitalk.asp

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


Want an incoming phone call to be routed via SIP URI dialing elsewhere for free? You can do that with an OBi. Want to have an incoming call be routed through another service provider to a different phone number? You can do that with an Obihai ATA. Want to be able to dial into your ATA from a cellphone and have the ATA call you back (great if you have free incoming minute plans) and get access to all of the same services you have on your ATA? You can do that with an Obihai ATA's auto attendant feature: https://www.obitalk.com/forum/index.php?topic=66.0

Is your mom is calling you on your FPL number, but you want to route that call to your aunt in the U.S. for free using Google Voice? Or maybe you want that call to be routed to your uncle in Sweden using VoiP.ms' value rates. You can do that with an Obihai ATA.


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.


Ooma is selling a proprietary device and a single service (Ooma's).
Their SIP servers are located in California. 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.

This Obihai ATA is just the device, which allows you to use whatever VoIP/SIP service(s) you want.
You still have to look into what service you want to use first. Some popular ones 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/

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.

3. For voip.ms, test to the closest server to you:
http://wiki.voip.ms/article/Choosing_Server

4. For Anveo (Montreal POP), test to sip.ca.anveo.com

5. For The Spout (Vancouver), test to ca.sipfrom.thespout.ca

6. For Callcentric (New York), test to ping.callcentric.com

7. For Ooma (California), test to myxprov.ooma.com

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 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"

The only point I would make to Paul in Connecticut, who appears to have returned to cable phone service, is that surely there's less lag for him using Callcentric or VoIP.ms' New York SIP server. Connecticut is closer to NY than it is to California. It's not as though Ooma is the only other VoIP service available to him.

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.

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) Try firmware 3.1.0 (Build: 5264): http://fw.obihai.com/OBi202-3-1-0-5264.fw (also works with OBi200)
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.


(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


In your call status page, during T.38 protocol fax transmission, you'll see the following:
"Audio Codec = tx=; rx=G711U"
(which doesn't specifically state T.38, but this is the only indication OBi seems to provide)

If you're not using T.38, you'll see "Audio Codec = tx=G711U; rx=G711U" instead, which just means you're using the G.711u codec only.

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/crackers/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?

For Freephoneline, Follow Me in your Freephoneline web portal must be disabled (unless you route MTelemarketers somewhere where the call is picked up immediately) for call blocking via your Obihai ATA to work. Login at https://www.freephoneline.ca/followMeSettings and check your Follow Me settings.

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.


A. 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.


B. Navigate to Voice Services-->SP(voipservice) Service-->X_InboundCallRoute

add {(MTelemarketers):}


Here's an example of what an X_InboundCallRoute might look like For an OBi202, this would look like M, by the way, stands for Digit Map.

If you don't know what yourauthusername is, navigate to Voice Services-->SP(voipservice) -->SIP Credentials-->AuthUserName


C. submit/save/reboot

D. Navigate to User Settings-->User Defined Digit Maps

i. Pick an unused User Defined Digit Map

ii. For the Label, enter Telemarketers

iii. For the DigitMap, enter phone numbers you want to block.

For example, (1234567890|4168888888|5193333333)

E. Submit/save/reboot

Note that you must enter phone numbers as they appear in your VoIP service's call log. For FPL users login at https://www.freephoneline.ca/callLogs


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 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
or (for sip calls) 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. Maybe you want to send them to Lenny: http://toao.net/595-lenny (keep in mind that sending telemarketers to Lenny will let telemarketers know your phone number is active).

You can also create a White list: newegg-obihai-obi200-59-99-a-1825095/4/#post23792781

Good luck!

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)

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 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)

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.

Updated

Getting one way audio issues with an OBi200/202 and Freephoneline? Are incoming calls not ringing? Can you not hear one side of the conversation (you can hear the caller, but the caller can't hear you or vice versa)?

These instructions do not address "this account is not valid" messages (you would need to contact FPL/Fongo for that problem).
"If you’re getting an “invalid account” error messages, or if people trying to call you are hearing "invalid account" or a busy signal, please log in to your account online at https://www.freephoneline.ca/followMeSettings and reset your Follow Me settings (or disable it). Please ensure your temporary FPL number is not listed as one of the Follow Me numbers."

If you have calls going straight to voicemail, login at https://www.freephoneline.ca/voicemailSettings and ensure "Rings before voicemail" is greater than 1. Also, check in your ATA to ensure you don't have "Do Not Disturb" enabled. This is found after logging into your ATA or at Obitalk.com under Voice Services-->SP(FPL) Service-->Calling Features-->DoNotDisturbEnable. Ensure there is no checkmark under "Value".
Navigate to Voice Services-->SP (FPL) Service-->Calling Features
a) Ensure DoNotDisturbEnable is unchecked
b) Ensure CallForwardUnconditionalEnable is unchecked
c) Ensure CallForwardOnBusyEnable is unchecked
d) Ensure CallForwardOnNoAnswerEnable is uncheked
e) Ensure AnonymousCallBlockEnable is unchecked


Often the problem is due to RTP packets not reaching the ATA. Common causes involve poorly functioning SIP ALGs (especially true with certain Netgear routers) in routers or NAT firewalls.

Hardware Specific Issues

A. Netgear R7000 routers

Disable SIP ALG in this router. Then reboot modem, router, and ATA in that order. Then test again.

If you have a Netgear R7000 router, you may need to install XWRT firmware. Afterwards, turn off the router and the ATA. Turn on the router. Wait for it to be fully up and running (including wi-fi). Then turn on the ATA. Download XWRT-Vortex here: http://xvtx.ru/xwrt/download.htm

B. Nettis 4422 modem from Carry Telecom (click the "Internet" tab)
http://www.carrytel.ca/support.aspx
Q : DSL - My VoIP phone does not work with Netis 4422 modem.
A : Please download the newest Netis firmware at www.carrytel.ca/download/netis1228.zip. Unzip the netis1228.zip file and update the firmware file netis1228.img for your modem. The new firmware has been tested and working with most of Voip phone providers

C. Hitron CGN3 series modem/router combos from Rogers
Typically it's better to have your own router and to stick whatever modem/router combo your ISP gives you into bridge mode. Otherwise, get Rogers via @CommunityHelps or TechXpert to disable SIP ALG for you. The TechXpert you speak to may not know how to disable SIP ALG. Be prepared for stupidity and resistance. Someone may try to enable DMZ in your Hitron modem/router; that's a security risk and very stupid. Be aware if you reset your modem or when Rogers pushes new firmware to your Hitron modem/router combo, SIP ALG will be enabled again by default, but there's a workaround listed in the second quotation below.



http://communityforums.rogers.com/t5/fo ... 972#M28972

Datalink wrote: Call tech support and ask the CSR to disable the SIP/ALG setting of the modem.

If the CSR refuses to check or uncheck the function switch, or tries to direct you to the Techxpert support which is a pay service, terminate the call and send a private message to @CommunityHelps to disable the SIP/ALG. Include your modem MAC address and Cable Account Reference Number in the text area. The Cable account reference number is located within the Internet section of your bill. If you are a new customer, you will not have immediate access to the Cable Account Reference Number. This can be obtained by calling Customer support. You can then send that Reference Number, along with the modem MAC address to CommunityHelps. The account number that you normally see or use is comprised of various home services such as Internet, Home Phone, Home Monitoring, etc, but the requested reference number is located at the top of the Internet section of your monthly account statement.

The modem MAC address can be found on the sticker at the back of the modem, or in the HFC MACC Address located in the Status page of the modem when you are logged into the modem.


When that is confirmed as complete, reboot the modem to determine if disabling the SIP/ALG has remedied the problem.

http://communityforums.rogers.com/t5/fo ... d-id/12535

Datalink wrote:The only problem now is that a modem reset will require you to send a pm to @CommunityHelps to disable the SIP/ALG setting again. To possibly avoid that, do the following:



1. Login into the modem,

2. Navigate to the ADMIN..... BACKUP page.

3. Run the Backup function and store the backup configuration file somewhere on your pc.



If you ever have to reset the modem, return to the ADMIN.... BACKUP page, run the Restore function using the configuration file that you have just created and then reboot the modem. Hopefully that also restores any parameters set by @CommunityHelps, which you are unable to accomplish from the user interface

For everyone with one-way audio issues, follow these steps:

i. Before beginning the steps below make sure whatever modem/router combo your ISP gave you is in bridge mode if you are using your own router. Call/contact your ISP if you have to.


1. Disable any and all port forwarding and/or DMZ in your router. Port forwarding creates security issues and can open the door to SIP scanners and hackers. If you're having trouble with SIP scanners and/or telemarketers, visit newegg-ca-obi200-39-99cad-obi202-59-99c ... st24563087

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.

(In Obitalk.com, you will need to enable and enter expert settings to do the following, if you want to use Obitalk.com. You do this by selecting Edit Profile-->Advanced Options-->check Enable OBi Expert Entry from Dashboard-->submit))

Keep in mind too, that if you're using the Obitalk.com web portal, after you submit a new setting, it take several minutes before Obitalk.com pushes the changes you've made to your ATA. Your ATA should reboot automatically after the changes are submitted.


3. In your Obihai ATA or at Obitalk.com, Navigate to Voice Services-->SP(FPL) Service-->X_UserAgentPort
Pick a random number between 30000 and 60000

(submit/save/reboot)

4. Navigate to Service Providers-->ITSP Profile (FPL)-->SIP

i) ensure X_DiscoverPublicAddress is enabled (it is by default)

ii) enable X_UsePublicAddressInVia (it's not by default)
You will need to uncheck default, device default, and Obitalk settings boxes. Then check the box to enable the feature

(submit/save/reboot ATA)

5. Retest
When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.

Then retest by calling your FPL phone number. If the problem is solved, don't continue.

6. If that doesn't work, you can also try enabling X_DetectALG (Navigate to Service Providers-->ITSP Profile (FPL)-->SIP)

(submit/save/reboot ATA)

7. Retest
When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.

Then retest by calling your FPL phone number. If the problem is solved, don't continue.

8. If that still doesn't work, disable X_DetectALG. And submit/save/reboot ATA.

9. If there are still problems, try disabling the SIP ALG feature in whatever router or modem/router combo it is that you're using:
http://www.obihai.com/faq/sip-alg/calling-out
I'm of the opinion Apple routers don't offer this feature, but you might as well check. If you manage to disable SIP ALG in the router, then retest.

DLINK router users may need to log into the admin page of their router, click the "Advanced" tab and then "Firewall Settings",
navigate to "Application Level Gateway (ALG) Configuration", and uncheck SIP: http://www.support.dlink.com/emulators/ ... dv_dmz.htm

If you received a modem/router combo, from your ISP ask your ISP. It is typically better to stick the modem/router combo from your ISP in bridge mode and use an external router.

See here for an example on how to disable SIP ALG in a router: http://www.obihai.com/faq/sip-alg/disable-alg



Save settings.
Turn off both router and ATA. Turn on router. Wait for router to be fully up and transmitting data. Turn on ATA.
Then retest by calling your FPL phone number. If the problem is solved, don't continue.

10. Try Proxyserver voip4.freephoneline.ca:6060
visit http://forum.fongo.com/viewtopic.php?f=15&t=16196 (look at the .pdf)
Make sure you refer to step 2 again.
(I'm of the opinion that step #6 makes this step redundant, but trying this is worth a shot anyway).
Retest. When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.

Then retest by calling your FPL phone number. If the problem is solved, don't continue.

voip4.freephoneline.ca:6060 is a SIP server whose purpose is to help those with SIP ALG issues (can't disable it in the user's router, for example).

So steps #6, #9, and #10 are all related. They are attempts to address a problem created by SIP ALG.


11. Try this at your own risk: use voip3.freephoneline.ca as the proxyserver
Make sure you refer to step 2 again.
voip3.freephoneline.ca is intended for testing purposes only--or for those who receive explicit permission to use it. Using it for an extended period may get your account banned. However, if using voip3.freephoneline.ca does work, you should open up a ticket with support and let them know that you can't get two-way audio any other way: https://support.fongo.com/anonymous_requests/new

If no responds to your support ticket, provide the ticket number in a private message to Fongo Support: http://forum.fongo.com/ucp.php?i=pm&mode=compose&u=7852

FPL configures its SIP servers differently than many other VoIP providers.
voip3.freephoneline.ca conforms more to the norm. But using it without permission can get your account banned.
If you'd like to avoid getting your account banned, use Proxyserver voip.freephoneline.ca, voip2.freephoneline.ca, or voip4.freephoneline.ca:6060 instead and skip to step #14.

12. Retest. When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.

Then retest by calling your FPL phone number.

13. If none of that helps, then, unfortunately, you're pretty much stuck with port forwarding your RTP (UDP) port range 16660-16798 from your router to your ATA. For reference, that range can be found under ITSP Profile (FPL)-->RTP. Then look at LocalPortMin and LocalPortMax. RTP packets need to reach your ATA in order for you get incoming audio. Quite often, when the one way audio issue occurs, this is the problem. RTP packets are not reaching your ATA. Ideally, one should not have to port forward in order to achieve proper two-way audio, since port forwarding does create security issues. Port forwarding should only be done when everything else fails.

Refer to the port forwarding section of your router manual to learn how to port forward to your ATA. If a router was given to you by your ISP, call your ISP.

14. Retest. When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.


Then retest by calling your FPL phone number.

15. If all else fails, try posting at http://forum.fongo.com/viewtopic.php?f= ... &start=300 and/or open a support ticket at https://support.fongo.com/anonymous_requests/new
If no responds to your support ticket, provide the ticket number in a private message to Fongo Support: http://forum.fongo.com/ucp.php?i=pm&mode=compose&u=7852

When I say Retest, retest always includes the following: A. Turn off both router and ATA. B. Turn on router. Wait for router to be fully up and transmitting data. C. Turn on ATA.

How do I stop Obihai from updating firmware without my permission?


If you don't disable Obitalk Service (in addition to other settings), Obihai can remotely upgrade firmware on your device without your permission, in the middle of a storm with power flickering, while you're out somewhere. If the power goes out in the middle of a firmware upgrade, you run the risk of wrecking your ATA. Do you want a bricked device? I don't. A UPS might be a good idea, by the way (http://www.newegg.ca/UPS/SubCategory/ID-72. Doesn't have to be from Newegg. Tons of places sell them).

If firmware can be upgraded remotely, Obihai has access to your ATA and can make changes. That's fine if you need technical support. It's not fine if you're into privacy.

After 1 year is up, in order to continue using Obitalk.com to upgrade firmware, you'll need to pay a $10 USD annual fee. However, you don't need to pay if you upgrade firmware manually. Best to learn now how to do it manually, unless, of course, you like paying annual fees.

There's two ways to update firmware for free even if the ATA is out of warranty:

A. Dialing ***6 (and then pressing "1" if an update is available)

http://www.obihai.com/docs/OBiProvisioningGuide.pdf (page 15)

This is a faster method than using the Obitalk web portal since you don't have to log into anything.

B. Manually updating firmware via the device:
http://www.obihai.com/docs/OBiDeviceAdminGuide.pdf (page 43)

Also found after visiting http://www.obihai.com/docs-downloads and clicking on "OBi Device Firmware"


New firmware releases offer new options that sometimes aren't quickly added to the web portal. Consequently, imo, it's best to stop relying on the web portal. You can (and, imo, should) use the Obitalk.com portal for initial setup (and keep in mind that you must use the web portal to activate Google Voice on your ATA), but eventually, people should learn to stop using it, in my opinion--unless, of course, you want to use OBiExtras ($4.99/month USD service). I don't.


To ensure Obihai and a ITSP provider can't, without your permission, update your device's firmware remotely, you must do the following (all three steps):

1. Dial ***1, and enter the IP address you're told into your web browser

2. Navigate to System Management-->Auto Provisioning

a) For Auto Firmware Update ---> ensure method is disabled

b) For ITSP Provisioning ---> change method to disabled (uncheck default box)

c) For OBiTalk Provisioning--->change method to disabled (uncheck default box)
Keep in mind changes made in Obitalk.com web portal will no longer transfer to your device
after this change is made.

(submit)

3. Navigate to Voice Services-->OBiTALK Service-->Obitalk service Settings
a) uncheck enable and uncheck the default box
You will not be able to call other Obihai devices using their respective Obitalk numbers after making this change. Also the OBi Echo test will no longer work if you disable OBiTALK Service.

(submit/reboot)

Credit goes to Pianoguy for teaching me about this issue.



A lot of related information and question/answers can be found over here: newegg-ca-obi200-39-99cad-obi202-59-99c ... k-1882793/

Sweet! This is like a databank for OBi202 users. Thanks @Webslinger

Well done Webslinger!

How Do I Setup VoIP.ms (or another service) as a Voice Gateway? (for outgoing calls only)

@puska83,

You were asking me about voip.ms in PMs, and I brought up setting up this service as a Voice Gateway instead of using up a SP trunk slot.

So, I'm going to respond here in case anyone else wants to learn.

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

puska83 has FPL setup on SP2. This is important to keep track of. For a Voice Gateway to work, another SIP 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.

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.)


1.
a. Navigate to Voice Services-->Gateway and Trunk Groups
b. Select an unused Voice Gateway
c. Enable check
d. Name VoIP.ms
e. AccessNumber is SP2(toronto.voip.ms) or SP2(whatever sever you have specified in your voip.ms web portal)
d. DigitMap (<*2:>XX.)
You can change *2 to whatever you want. So, when you dial *2 before the phone number, it goes out over VoIP.ms
XX. stands for any number you can punch in.

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.

e. AuthUserID your voip.ms userid
f. AuthPassword your voip.ms password

(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|)

Again, you just need to match *2 with whatever you used in step 1d.

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 a 3 second delay before the phone number is sent by your ATA.

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

c. You need to add

{(Mvgx):vgx}

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:
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 "1".

d. submit/save/reboot

What's the easiest way to setup a VoIP service with an Obihai ATA?

BestFind wrote: ↑ I have now received the Obi202 that I bought and have also received the VOIP unlock code from FPL. What now? Do you have step by step instruction onto how to configure my Obi202 with FPL?

I would recommend posting in this thread instead for FPL issues: merged-freephoneline-ca-free-local-soft ... ip-821229/

I do recommend using the Obitalk portal for initial setup until newcomers get things working the way they want to, especially if they plan on using Google Voice. After you're comfortable and setup properly, stop using the portal (refer to newegg-obihai-obi202-voip-phone-adapter ... st25148699 if you want to stop using the portal).

1)The easiest way to setup a VoIP account on an Obihai ATA is to visit http://www.obitalk.com/obinet/
2)Then create an account/register.
3)Then add your Obihai ATA to the portal.
4)Then click the SP# (for FPL use SP1) you wish to configure.
5)Then scroll down the page and select the "next" button for "Obitalk Compatible Service Provider" (you will have to pay for a VoIP service, unless you have a Google Voice phone number).
6)Click "Accept" on the 911 pop up window.
7)Select the appropriate VoIP provider from the list.
8)If you're using your VoIP service for 911 service, then ensure "Use This Service for Emergency 911 Calls" is selected.
10) If required, select the closest Service Provider Proxy Server to you (or chose the one you get the lowest pings/jitter from).
11)Fill in Username and Password (provided by your VoIP provider. With FPL, log in at https://www.freephoneline.ca/showSipSettings; SIP Usename and SIP password).
12) Click "save"

I was reviewing your post and saw that it seems that one can go thru obitalk or IP address but not both.

correct

Since obitalk.com seems to ask for $10 (eventually outside of my warranty period) I would want perhaps to start to do manually.

They only charge $10 USD for updating firmware via the Obitalk web portal outside of your warranty period (and if you to email or call them for technical support outside of the warranty period). That's it. There's two other ways of updating firmware for free.

You can still use the web portal to do whatever you want, after updating firmware, outside of the warranty period.

Can You give me a link that can guide me?

Visit https://www.obitalk.com/info/faq/OBi202 ... b-from-WAN

For FPL manual setup guides . . .

http://forum.fongo.com/viewtopic.php?f=15&t=16090
(For manual setup)

or

http://forum.fongo.com/viewtopic.php?f=15&t=16196 (use this if you're not using your own router; that is, if you were provided a crummy modem/router combo from your ISP then chances are this guide is going to work better for you)
(for manual setup)

Edit: Should you encounter one-way audio issues or dead air with FPL, visit merged-freephoneline-ca-free-local-soft ... st24955445