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freephoneline.ca - Free Local Soft Phone Line for lifetime VOIP

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NAT filtering "disable sip alg" checked, "Open" is check,should I go with "Secured" option?Another issue is when pick up the phone,I didn't get dial tone but noise sound like "di di di ",but phone still working,please advise.

This is for my co-work,she has Netgear N300 router I guess.
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rui wrote: NAT filtering "disable sip alg" checked, "Open" is check,should I go with "Secured" option?
Never having used the N300 or most Netgear routers with default firmware, I don't really know the answer to this question. It's impossible for me to become familiar with all routers on the market.
This thread was less than useful (lots of people guessing): https://community.netgear.com/t5/Genera ... td-p/39980.
I would hope that one of the options (either secure or open) would offer the user a Restricted Cone NAT or Port Restricted Cone NAT firewall.

Basically, I don't have an answer, except to test first.


When buying a router for VoIP, ensure you buy one 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. They’re fine.


So, I would run that test with both the "secure" and "open" options to see what the results are.
I do see someone at dslreports recommending to use "open" with the N300: http://www.dslreports.com/forum/r260447 ... N300-Setup.
But I would still run the above test first.

You do not want a Full Cone NAT router.

Also, if your co-worker is using DSL, the Netgear router should be doing the PPoE login. If you're co-worker is using cable, whatever modem/router combo or gateway her ISP gave her should be in bridge mode.
Another issue is when pick up the phone,I didn't get dial tone but noise sound like "di di di ",but phone still working,please advise.
Yeah, that sounds like there's a voicemail message waiting to me.

On the surface, it seems like Freephoneline is working perfectly fine for her. She may not have to change anything (test with an incoming phone call to her FPL number). If there's two-way audio, then great.
Last edited by Guest1284983 on Mar 15th, 2017 3:02 pm, edited 1 time in total.
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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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Webslinger wrote: Also, if you're co-worker is using DSL, the Netgear router should be doing to PPoE login. If you're co-worker is using cable, whatever modem/router combo or gateway her ISP gave her should be in bridge mode.
She is using CIK cable now but will be using carrytel cable 100M plan by the end of Apr.

Webslinger wrote: Yeah, that sounds like there's a voicemail message waiting to me.
On the surface, it seems like Freephoneline is working perfectly fine for her. She may not have to change anything (test with an incoming phone call to her FPL number). If there's two-way audio, then great.
I think so, will ask her to check,thanks so much.


[/quote]
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rui wrote: She is using CIK cable now but will be using carrytel cable 100M plan by the end of Apr.
Okay, the bottom line is if they gave her a modem/router combo, she should stick that device in bridge mode.
If they just gave her a modem without any router functions built into it, then she doesn't have to do anything.
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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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Firmware 3.1.1 (Build: 5589) is now the latest OBi200/202 ATA firmware: http://fw.obihai.com/OBi202-3-1-1-5589.fw

Changelog is located at https://www.ukvoipforums.com/viewtopic.php?f=25&t=987

If you're not having problems, there's no need to update.
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: The most powerful ATAs for home use are Obihai OBi200 or OBi202 (if you need two phone or FXS ports), due to their powerful call routing features, sold by Newegg Canada.

OBi200 is currently on sale: newegg-obihai-obi200-ata-49-99-1-50-ehf ... x-2083622/
Today (Sunday) is the last day of the sale.

Obihai ATAs were designed by the same people who engineered the PAP2T.
Thanks Webslinger for the response.

Unfortunately, I wasn't able to buy the OBi200 that was on sale because I was able to get my Linksys PAP2T going again for a few weeks and then it went stopped functioning again.

I have a question for anyone that can help.
My Linksys PAP2T like I mentioned stopped working.
It displays a red solid light for Power and a green solid light for INTERNET. There's no light on for Phone Line 1 or 2. Does anyone know what could possibly be wrong? Is it bricked?

One last question, what are the differences/advantages of buying a OBi200 over a Cisco SPA122, ATA with Router - Affordable and Feature-Rich Voice over IP (VoIP)?

Any help would be appreciated.

Thank you.
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fogo wrote:
It displays a red solid light for Power and a green solid light for INTERNET. There's no light on for Phone Line 1 or 2. Does anyone know what could possibly be wrong? Is it bricked?
Is it a fake? http://voipfan.net/other/fakepap2.php

I believe the red light indicates the boot loader isn't even working. There may be a poor soldering job.

You could also try popping it open to look for bulged capacitors (maybe the unit is also overheating). You could also try another power supply.

To me, the solid red light doesn't sound promising. I'm guessing you can't even get it to do a factory reset, but if you can that might be worth a shot.

One last question, what are the differences/advantages of buying a OBi200 over a Cisco SPA122

Given the price difference when an OBi202 goes on sale, any other 2 phone jack ATA for home use is a waste of money, imo. Why anyone would limit him or herself to something inferior over a difference of $20-$45+tax doesn't make any sense to me at this price point--despite Obihai having annoying support policies.

Also, visit http://www.dslreports.com/forum/r304704 ... -providers

If you absolutely need two phone ports, then I would look into an OBi202 at Newegg Canada instead of an OBi200.


Pianoguy is one of the biggest VoIP gurus on RFD (and in general):
PianoGuy wrote: The SPA's features are very rudimentary compared to the OBi which has many excellent inbound call routing features, as well as the ability to seamlessly use multiple providers for incoming and/or outgoing calls - including Google Voice - on a single line. I know you're trying to save a buck and I don't blame you for that - this IS RedFlagDeals after all. But when you think how much you'll save in monthly costs, the extra $61 ($76-15) for an [OBi] will seem pretty insignificant.
The first post in this thread (click) pretty much explains everything an SPA122 ATA can't do that an Obihai can, starting with "OBi200/202 offers support for up to 4 SIP trunks and 8 voice gateways. That means you can use up to 12 different VoIP services on one telephone . . ."

It's not just an advantage with Google Voice. It's also iNUM, SIP Broker, etc., and being able to route calls between them. It's being able to route telemarketing calls elsewhere as I've outlined in my guide with Freephoneline and being also able to use Nomorobo with Freephoneline, etc.

Obihai ATAs also have far more powerful call routing abilities.

They're almost mini-PBXes.
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: Is it a fake? http://voipfan.net/other/fakepap2.php

I believe the red light indicates the boot loader isn't even working. There may be a poor soldering job.

You could also try popping it open to look for bulged capacitors (maybe the unit is also overheating). You could also try another power supply.

To me, the solid red light doesn't sound promising. I'm guessing you can't even get it to do a factory reset, but if you can that might be worth a shot.





Given the price difference when an OBi202 goes on sale, any other 2 phone jack ATA for home use is a waste of money, imo. Why anyone would limit him or herself to something inferior over a difference of $20-$45+tax doesn't make any sense to me at this price point--despite Obihai having annoying support policies.

Also, visit http://www.dslreports.com/forum/r304704 ... -providers

If you absolutely need two phone ports, then I would look into an OBi202 at Newegg Canada instead of an OBi200.


Pianoguy is one of the biggest VoIP gurus on RFD (and in general):


The first post in this thread (click) pretty much explains everything an SPA122 ATA can't do that an Obihai can, starting with "OBi200/202 offers support for up to 4 SIP trunks and 8 voice gateways. That means you can use up to 12 different VoIP services on one telephone . . ."

It's not just an advantage with Google Voice. It's also iNUM, SIP Broker, etc., and being able to route calls between them. It's being able to route telemarketing calls elsewhere as I've outlined in my guide with Freephoneline and being also able to use Nomorobo with Freephoneline, etc.

Obihai ATAs also have far more powerful call routing abilities.

They're almost mini-PBXes.

Webslinger, thanks again.
I do appreciate your time and effort.
Your post is much appreciated.

fogo
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http://status.fongo.com/

"We will be performing an extensive upgrade to our backend voice calling system on March 28 and 29 from 12:00AM-06:00AM EST. Voice calling may be intermittently disrupted; please do not submit a support ticket during these times."
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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http://status.fongo.com/

"Voicemail notifications are currently not working. Voicemail is still being collected and received. Engineering is investigating. To check if you have new voicemail after a missed call dial *98. - 2017-03-28 16:27"
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: If you have an OBi200/202, you don't need to do the step you're trying to do. Instead, you should be enabling X_AcceptSipFromRegistrarOnly

You're doing a step that's intended for OBi100/110 ATAs.
Hi I'm having this issue again for another OBi200. I can't seem to find X_AcceptSipFromRegistrarOnly in the settings though. Has this step been changed to something else now? I'm having the issue where we get lots of calls from 4 digit numbers at random times. Thanks in advance!
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risktaker91 wrote: I can't seem to find X_AcceptSipFromRegistrarOnly in the settings though. Has this step been changed to something else now?
No. And if you follow this PDF guide, following each step, you won't have that problem: http://forum.fongo.com/viewtopic.php?f= ... 545#p74545



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.
voip2.freephoneline.ca is 162.213.111.22
voip4.freephoneline.ca is 162.213.111.21

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.

Note that unless your ATA is registered with voip.freephoneline.ca, Fongo Mobile calls to your Freephoneline phone number will be dropped straight to voicemail if X_AcceptSipFromRegistrarOnly is enabled. Fongo Mobile calls to Freephoneline phone numbers are SIP URI calls.


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

Code: Select all

{(MTelemarketers):},{>('yourauthusernamegoeshere'):ph}
For an OBi202, this would look like

Code: Select all

{(MTelemarketers):},{>('yourauthusernamegoeshere'):ph,ph2}
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)


Some people ask about blocking anonymous or unknown calls
(?|un@@.|Un@@.|anon@.|Anon@.)

? = no Caller ID
@ = any single alphanumeric (number or letter) except #
[mention2=329452 slug=].[/mention2] = zero or more occurrences of any length alphanumeric (number or letter) sequence except #
@@. = any single alphanumeric (number or letter) except # followed by zero or more occurrences of any length alphanumeric sequence except #

un@[mention2=329452 slug=].[/mention2] will catch unknown or anything starting with un followed by at least one more character (except #)
anon@. will catch anonymous or anything starting with anon (except #)


I do not generally recommend blocking anonymous calls since doctors and hospitals can call from them.

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

Code: Select all

{(MTelemarketers):sp2(phonenumbertosendtelemarketers)}
in FPL's X_InboundCallRoute in place of {(MTelemarketers):} to send those telemarketing calls to another phone number.

If FPL is SP1, you can also use

Code: Select all

{(MTelemarketers):sp1(phonenumbertosendtelemarketers)}

or (for sip calls)

Code: Select all

{(MTelemarketers):sp1(sipnumber@sipdomain.com)}
It doesn't really matter. But if you don't want telemarketing calls to drop straight to FPL's voicemail, it is possible with an Obihai ATA, to route these calls elsewhere. 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).

Update

I think this might be a better solution for Telemarketers for FPL users than what I posted previously.

Here are the steps I took:

1. Went to www.tropo.com
2. Created a free developer account
3. Verified account and logged in
4. Found an audio file that plays SIT tones followed by a "We're sorry, you have reached a number that has been disconnected..."
5. Clicked on "My Files" in Tropo and stuck the file in the www folder
6. Selected "My Apps" and clicked "create application"
7. Entered nogood for Basic information (you can put whatever you want here)
8. Clicked on "new script"

Entered the following:

Code: Select all

<?php
say("http://hosting.tropo.com/mytropoaccount#/www/disconnectedmessageaudiofilethatIadded.mp3");
say("http://hosting.tropo.com/mytropoaccount#/www/disconnectedmessageaudiofilethatIadded.mp3");
?>
9. Saved the script as nogood.php (just has to end with .php)

10. Clicked "create app"

11. Scrolled down and picked a free Tropo phone number for Canada

12. Stuck {(MTelemarketers):sp1(mytropophone#)} in X_InboundCallRoute for FPL in my OBi

(where SP1 = FPL), but it doesn't matter what SP you use, as long as you call your Tropo phone number for free using it.

Rebooted


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


How do I setup Nomorobo with Freephoneline (or any service provider that doesn't support Nomorobo) with an Obihai ATA?

So, I got Nomorobo working with Freephoneline, and really, with an Obihai ATA and a toll free number from access number from Nomorobo, you can use Nomorobo with any service provider.

Keep in mind, if you have Follow Me enabled in your Freephoneline web portal, that in order for Nomorobo to work properly, your Nomorobo access number has to ring simultaneously (or first if you're using sequential forwarding). Freephoneline doesn't allow toll free phone numbers to be used with Follow Me at the time of this writing. If you're having problems getting Nomorobo working with Follow Me enabled, disable Follow Me in your Freephoneline web portal account.

1. Sign up at https://www.nomorobo.com/

2. Select "Landline/VoIP" for your phone type.

3. Select Vonage as your carrier (or choose one that gives you a toll free nomorobo number). The bolded part is the important step.
I don't know if choosing other carriers will provide you with a toll free Nomorobo phone number, but maybe some will.

4. For phone number, enter your VoIP phone number that you want to protect with Nomorobo

5. Click next (and keep this webpage open)

6. Open another tab in your 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.

7. Navigate to Voice Services-->SP(voip service you want to protect with nomorobo) Service-->X_InboundCallRoute

Yours might look like {(yourcellphonenumber1|yourcellphonenumber2:aa($1)},{(MTelemarketers):},{(MLenny):spx(13475147296@in.callcentric.com)},{ph,spx(1866732xxxx@tollfree.alcazarnetworks.com;ui=$1)}

You just need to add the bolded part.
For OBi202s, you would use {ph,ph2,spx(1866732xxxx@tollfree.alcazarnetworks.com;ui=$1)}

The x in SPx needs to be a number between 1 and 4, where you've setup a SIP service provider. For example, if Freephoneline is setup on SP1, then use sp1.

1866732xxxx is your Nomorobo access phone number. You need to replace the xxxx.

Google Voice is not SIP. GV is XMPP with Obihai ATAs.
So, if you don't have a SIP service, you would need to setup a dummy SIP trunk as shown over here by azrobert:
http://www.obitalk.com/forum/index.php? ... 3#msg68413

You can also use sip.tollfreeproxy.com instead of tollfree.alcazarnetworks.com. I'm not sure, out of all the free SIP tollfree termination services available that don't require registration, which is the most reliable.

8. Save settings/reboot ATA

9. Test your nomorobo number. Go back to the other web browser tab with nomorobo. Select "I'm ready. Call me now." Or choose the test call option.

10. When your phone rings, answer it. If your phone doesn't ring, step 9 probably doesn't matter too much. See here: freephoneline-ca-free-local-soft-phone- ... #p27561047

That's it.


What happens with Nomorobo is that when the incoming caller ID isn't considered a telemarketer, Nomorobo responds with a busy signal, and your phone ports will still ring. If Nomorobo determines the caller is a telemarketer or robodialer, Nomorobo answers the call, and your phone ports stop ringing: https://nomorobo.zendesk.com/hc/en-us/a ... ks-a-Call-
Nomorobo may take a ring or two before answering the call (your phone will likely ring once per telemarketing call).


Keep in mind that, like Lenny, if Nomorobo answers the call, telemarketers will know your phone number is active and possibly add your phone number to other calling lists. So, in my opinion, using Tropo with SIT tones in combination with Nomorobo is the best solution. When Nomorobo answers the call, make note of the incoming phone number and add it to your MTelemarkers Digit Map.


Good luck!
Last edited by Guest1284983 on Apr 23rd, 2017 11:43 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.
Jr. Member
Dec 4, 2007
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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.

I don't see this in my settings. I have posted a screenshot of what I see below.

http://imgur.com/a/yjG7A
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risktaker91 wrote: 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.

I don't see this in my settings. I have posted a screenshot of what I see below.

http://imgur.com/a/yjG7A

That setting has existed in Obihai 200/202 ATAs ever since firmware release 3.11.4581, and the latest version is 3.1.1 (Build: 5589): http://fw.obihai.com/OBi202-3-1-1-5589.fw. That setting also exists at Obitalk.com

More importantly, that setting is not needed to block SIP scanners if you use a good NAT firewall and don't port forward. You can also use X_EnforceRequestUserID. You don't need to use X_AcceptSipFromRegistrarOnly.

The setting is shown on page 22 of the PDF guide: http://forum.fongo.com/viewtopic.php?f= ... 545#p74545. The setting is not enabled in the PDF guide because it's not needed and can also cause issues with incoming calls from Fongo Mobile when the ATA is not registered with voip.freephoneline.ca. But you can enable that setting if you want, provided you have an Obihai OBi200 or OBi202 and are using firmware build 3.11.4581 or later.

Those settings do not exist for OBi100/110 ATAs.

Note that by upgrading firmware above 3.0.1.4350 (pre-OAuth 2.0 firmware) you will lose the capacity to configure Google Voice locally via the ATA. Instead, you will be forced to use Obitalk.com for Google Voice.
Last edited by Guest1284983 on Apr 23rd, 2017 2:57 pm, edited 3 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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sunnyhappy wrote: setting up Free phone line is using OBI Dashboard Configuration.
Section 11:
Alternatively, you may use the OBi Expert Configuration page via the OBiTALK web portal to enable WAN
access.
- Sign in to www.OBiTALK.com (link)
- Select the OBi202 from the OBi Dashboard
- Go to: System Management - Device Admin
- In the Web Server area, you may enable AccessFromWAN by checking the available box.”

The OBITalk.com is not allowing me to select the accessfromWAN after I select DeviceDefault flag ^ or password, it is not allowing any change on this page.
You need to uncheck both the "device default" and "OBiTALK settings" boxes or the right hand side before you can check "AccessfromWAN" (and then you need to click the "submit" button below).
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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@sunnyhappy

Also, if you're using Obitalk.com, you can skip steps 3 and 4 (on page 12). You're, based on what you've written, using Obitalk.com and not logging in via the local IP of the ATA.

If you have Google Voice configured on SP1, then you'll need to configure FPL on SP2 (and ITSP Profile B).

ITSP Profile A corresponds with SP1.

And then under Physical Interfaces-->Phone Port 1 (and possibly Phone Port 2, if you're using both phone ports), you'll want to change your Primary Line to whatever FPL is set to (SP2 service, possibly).
It's usually better to have your outgoing calls go through FPL so that your Canadian phone number is shown whenever possible.

Then, if GV (Google Voice) is setup on SP1, you would dial **1 before the phone number when you want to dial through Google Voice.

At default settings, it makes more sense to have FPL configured on SP1 and GV configured on SP2 instead, since you don't have to fiddle with Primary Line settings.
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.
Newbie
Oct 23, 2008
69 posts
21 upvotes
Toronto
The FPL is working , but now GV is not working, both are on Same phone line 1 on OBI202 box

used manual to set up the config file using OBITalk Dashboard setup

SP1Google Voice Register Failed: No Response From Server
SP2 (Optional)freephoneline.ca Registered

There is only one configuration does it apply for both GV and FPL?

Is there a way to select the phone line for GV or FPL for outgoing calls from the phone before dialing?

In GV setp
Incoming Calls Will Ring OnPhone 1 only
Last edited by sunnyhappy on Apr 24th, 2017 9:57 pm, edited 1 time in total.
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sunnyhappy wrote: SP1Google Voice Register Failed: No Response From Server
Do this: http://www.obitalk.com/forum/index.php? ... 0#msg56460 (except for the 911 stuff)

Basically delete GV from your Obitalk account, and then add it back to SP1.

Be aware that both Obitalk service and provisioning must be enabled when provisioning GV: newegg-obihai-obi200-ata-49-99-1-50-ehf ... #p27125797 (read through my posts to canadaodyowner on that page).

There is only one configuration does it apply for both GV and FPL?
Sorry, I don't understand your question. If you're asking about the FPL PDF guide, the guide only applies to FPL.
Is there a way to select the phone line for GV or FPL for outgoing calls from the phone before dialing?
Under Physical Interfaces-->Phone Port 1 (and possibly Phone Port 2, if you're using both phone ports), you'll want to change your Primary Line to whatever FPL is set to (SP2 service).
By default, all calls will go out over the primary line (FPL, in this example).

If you want to use GV, by default you dial **1 (plus the phone number) to dial out on SP1.
**2 is for SP2
**3 is for SP3
**4 is for SP4

You do not need to dial **x, where x is a number between 1 and 4, for whatever service provider the primary line is set to.
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.
Newbie
Oct 23, 2008
69 posts
21 upvotes
Toronto
added the GV using the Icon on the Obitalk dashboard and it added as SP3 and connected, removed the SP1 GV and it is working.
Only issues is when I want to dail outgoing calls using GV , I first dial **3 then US number, IS there a short cut or this is the only was.
Want to use FPL as default calling out line for calls within canada.
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sunnyhappy wrote: Only issues is when I want to dail outgoing calls using GV , I first dial **3 then US number, IS there a short cut or this is the only was.
Click this.

If you're looking at what X360 did, you'll need to use {(Mcanada):sp2},{(Musa):sp3}


M stands for digitmap

If you want to just press 1 to have all calls go through GV, visit freephoneline-ca-free-local-soft-phone- ... #p24412847
Step 4 from that link would be
change {(<**2:>(Msp2)):sp2},{(<**3:>(Msp3)):sp3}
to
{(Msp2):sp2},{(Msp3):sp3}
or just add {(Msp2):sp2},{(Msp3):sp3} in front of/to the left of {(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2}


Again, under Physical Interfaces-->Phone Port 1 (and possibly Phone Port 2, if you're using both phone ports), you'll want to change your Primary Line to whatever FPL is set to (SP2 service).
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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