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Obihai OBi202 ATA $69.99 + $1.50 EHF+ free shipping+tax

  • Last Updated:
  • Jul 5th, 2020 10:23 am
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Mar 3, 2002
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I've heard scary stuff about VoIP 911. Isn't it unreliable?

VoIP E911 is a two step process. With Freephonline, after dialing 911, the initial E911 call centre, which does have my name, address, and call back number, still has to transfer the call to local dispatch (PSAP), which doesn't have my name, address, and phone number.

It's important, when signing up to a VoIP service you're planning on using 911 with that you always keep your address updated on file with them. If you move, update your address. Your VoIP service sends that information to the E911 call centre/Northern911, which they will keep on file.

In some rare instances, I suppose it's possible that Northern911 (I'm guessing this is what FPL and other VoIP services in Canada use, but I'm not sure) may not transfer to the correct local dispatch (PSAP) number (human error happens). Some people I configured services for in the past were very paranoid about VoIP E911 and forced me to do a test call. Worked fine. That is, the first person I reached had name and address info; they ask for confirmation. And the call was promptly transferred to local dispatch and correct address info was given to local dispatch, verbally, by the first call centre. Worked fine each and every time I was asked to test.

How does this compare to 911 with a landline?

Landline 911 is not a two-step process. You don't need to keep your address updated. Landlines are the most reliable for 911 calls.
But landlines don't work after your telephone lines have been knocked out by a storm.

How does this compare with Mobile 911?

Mobile 911 is not a two step process. However, they do not have your exact address, but they should have an approximate location (they should at least have the cellular site/tower that's carrying your call), especially if you're in a major city (they may have latitude and longitude). If you're in a rural area, location based on cellular towers may not be very precise. 70%+ of 911 calls are now coming from mobile phones according to the CRTC. Going forward, this is where improvements are going to be made.


Also, keep in mind that with FPL each E911 call is $35. If you dial 911 less than twice a year instead of paying $1.50 USD/month with Callcentric, Anveo, or VoIP.ms, you're ahead with FPL. And you're paying an ongoing minimum monthly fee of $3.98 with Ooma. Ask yourself how often you're calling 911. If you're a senior citizen with a lot of health issues, maybe FPL is a bad idea. (And I don't mean to belittle this point. Everyone gets old. Health is a serious matter.) Otherwise, you'll end up way ahead using a FPL in the long run (in terms of cost).

Here's the thing . . . I used to talk to FPL reps several years ago over the phone, back when they allowed tech support calls. And even then a e911 fee was listed (but not in the FAQs), and I inquired about it. I was told the fee was intended to dissuade people from test calling 911--and that people wouldn't actually be charged.

Fast forward to now, and the $35 per call E911 fee is listed in the FAQs. It's listed all over the place. It's certainly enough to prevent me from testing 911 on FPL. Reps are now saying you will be charged no matter what when you dial 911. Is that true? Maybe. Is that enough to scare me from testing 911? Sure. Has anyone been charged yet? I don't know. Anyway, no one is going to be calling 911 using FPL unless it's really necessary now, and if that's the intent, I'm fine with it. And if I really need E911 as a backup (my smartphone is always nearby), it's there for me. In the meantime, I'm not paying ongoing monthly fees for something I'm not using.


freephoneline-ca-free-local-soft-phone- ... #p27964332
__wizard__ wrote: As a customer with FPL, I used 911 service 3 months ago and never got the $35 charge
YMMV (your mileage may vary)


Obihai OBi200/202 ATAs with the OBiBT adapter can be paired with smartphones over bluetooth: http://www.obihai.com/obibt.
Then with an Obihai OBi 200/202 ATA, you'd add {911:bt} in your OutboundCallRoute, and then all of your 911 calls on your phones go out over your smartphone's 911 cellular service, provided your smartphone remains within bluetooth range of the ATA.


By the way, There's also Anveo's E911 service ($25 USD per year) available through the Obitalk.com web portal, as an alternative 911 service (limited to a maximum of 5 e911 calls per year): https://www.anveo.com/e911obi.asp (click the link for more information). People asking for help with this Anveo E911 service should probably ask canadaodyowner, who is using this service and is also a Freephoneline customer: freephoneline-ca-free-local-soft-phone- ... #p24980477. I have no experience with Anveo's special E911 service.


VoIP E911 is available all the time under these conditions:

1) You have electricity. A UPS is always a good idea.

2) Your internet service isn't out.

3) Your VoIP service isn't down.

I don't know anyone who doesn't have a smartphone.
Last edited by Guest1284983 on Dec 18th, 2018 10:44 am, edited 1 time in total.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
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deal link is wrong (it was from your session i believe.

Correct link is:
obi202

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


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

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

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


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

In this example, A SIP service (VoIP.ms, Anveo, FPL, etc.) is setup on SP2. This is important to keep track of. For a Voice Gateway to work, another SIP Service/trunk must be defined and established. Google Voice doesn't count because it's XMPP. Keep this in mind when you get to step 1e.

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

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



So, I would suggest doing something like this:

If you used the Obitalk web portal (www.obitalk.com) to configure your ATA, keep in mind that you must continue using it to configure your ATA. Otherwise whatever settings you change will eventually be overwritten by what you previously entered at obitalk.com anyway. If you wish to disable this behaviour, dial ***1. Enter that IP address into a web browser. Navigate to System Management-->OBiTalk Provisioning-->select Disabled for the method. Save. Reboot ATA. Now obitalk.com won't overwrite whatever changes you make via the device's interface (via IP address).

Pick one method (obitalk.com) or the other (IP address of device) for changing device settings. But do not use both methods. Keep in mind that activating Google Voice requires using the Obitalk.com web portal.

(In Obitalk.com, you will need to enable and enter expert settings to do the following, if you want to use Obitalk.com.)


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

Also ensure that X_DiscoverPublicAddress is enabled.

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



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

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

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


d. DigitMap (XX.)

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

XX. is an indefinite variable, which basically stands for anything you could possibly dial.

I'm just explaining this now for when you get to step 2a below.

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

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

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

(submit/save/reboot)

2. Navigate to Physical Interfaces-->Phone Port-->DigitMap

a. Add *2XX.S3 to your digitmap (this entry needs to be separated by "|", so use |*2XX.S3|)

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


((Mop)|[1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**8(Mbt)|**9(Mpp)|(Mpli)|*98|310xxxx|1xxxxxxxxxx|[2-9]xxxxxxxxx|*2XX.S3|**0|***|222222222|**9(Mpp))

Don't copy and paste this. You just need to add the bolded section. That's not even what mine looks like. I'm just providing an example.

XX. is an indefinite variable, which basically stands for anything you could possibly dial.
Because it's an indefinite variable, it's also subject to a 10s interdigit timeout, unless you specify the number of seconds you want your OBi ATA to wait for you to finish punching in a number.

So, XX.S3 would mean there's basically a 3 second delay while the ATA waits for you to punch in the phone number.

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

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

c. You need to add

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

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

x = the # of the voice gateway you chose in step 1b.
So change x to the number of the voice gateway you chose in step 1b.
M = digitmap

The outboundcallroute is processed from left to right.

So, if FPL is setup on SP2 and GV is setup on SP1, you might have something that looks like this:

{([1-9]x?*(Mpli)):pp},{(<#:>):ph2},{**0:aa},{***:aa2},{(<**2:>(Msp2)):sp2},{(<**1:>(Msp1)):sp1},{(<*2:>(Mvg2)):vg2(>012345*$2)},{(<**3:>(Msp3)):sp3},{(<**4:>(Msp4)):sp4},{(<**8:>(Mbt)):bt},{(<**9:>(Mpp)):pp},{(Mpli):pli},{011xx.:SP2},{911:sp2},{933:sp2},{([1-9]x?*(Mpli)):pp},{(<##:>):li},{(<#:>):ph2},{(<**70:>(Mli)):li},{(<**82:>(Mbt2)):bt2},{(<**81:>(Mbt)):bt},{(<**8:>(Mbt)):bt},{**0:aa},{***:aa2},{(Mpli):pli}

Just need to add what's in bold after whatever you have for sp2 and sp1. Don't copy and paste this in it's entirety. Just look at what I have in bold as an example. And 1 is the number of the voice gateway you chose before. So you may need to change the "2". You'll need to change 012345 to the 6 digit prefix you're actually using with Anveo Direct.

d. submit/save/reboot

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




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

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

https://forums.redflagdeals.com/google- ... #p29271035
Last edited by Guest1284983 on Apr 29th, 2018 12:35 am, edited 2 times in total.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.
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May 23, 2008
2591 posts
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Vaughan, Ontario
Webslinger wrote: Google Voice ends XMPP support in June '18 (Google Voice Gateway and Obihai hacked firmware users potentially affected)

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

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

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

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

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

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

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



That includes Freephoneline!
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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:( Tropo is shutting down: http://blog.tropo.com/2018/10/01/tropo- ... perations/.

Webslinger wrote:

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 #
@ . = zero or more occurrences of any length alphanumeric (number or letter) sequence except # (note that there should be no space after the @ symbol; I had to put it there on this forum)
@@ . = any single alphanumeric (number or letter) except # followed by zero or more occurrences of any length alphanumeric sequence except # (note that there should be no space after the @ symbol; I had to put it there on this forum)

un@@ . will catch unknown or anything starting with un followed by at least one more character (except #)
(note that there should be no space after the @ symbol; I had to put it there on this forum)
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:
freephoneline-ca-free-local-soft-phone- ... #p21660123
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: https://forums.redflagdeals.com/freepho ... #p21807239 Edit: It appears that OBiPlus has been completely killed by Obihai now.



Alternatively, 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.

That's it.

If you don’t receive the Nomorobo verification call, you may not have to worry.

For example, I’ve received the following question from another Freephoneline user before:

“I tried following your guide to get Nomorobo working with Freephoneline, but for some reason when I dial number I got from website it rings once and then I get a busy signal (okay, that’s what should happen). But when I click on `I'm ready. Call me now’, my home phone never rings.

I've tried alcazar... Have Follow me - Disabled on FPL

Here is my Inbound route for an OBi202:

{(MTelemarketers):},{ph,ph2,sp1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=$1)}

Any ideas what could be my issue?”


A few thoughts occur to me.

A) Possibly Nomorobo is blocking the outbound call due your phone number not being owned by Vonage (or the provider you selected to obtain the toll free Nomorobo access number). If you don't see the test call in https://www.freephoneline.ca/doGetCallLogs, that might be one explanation. However, that's not a perfectly reliable method for determining what's happening because incoming phone numbers only appear in FPL's call log if the incoming call has been answered by you or FPL's answering machine.

B) If the test call to your FPL number is a SIP URI call from Nomoboro, the test call will never reach you because FPL blocks incoming SIP URI calls unless they're from Fongo Mobile.

C) Possibly FPL doesn't allow incoming Nomorobo verification calls anymore.

D) This is the most important point: it doesn't seem to really matter if you can't receive the Nomorobo verification call. If you don't verify the test call, Nomorobo still picks up the call with "please try your call again" when a telemarketer calls. If the test call is verified, the telemarketer hears this: https://soundcloud.com/nomorobo/nomorob ... er-captcha. I might actually prefer if the test call isn't verified.

Regardless, I suspect if you are seeing "busy" in your Obihai ATA’s call history, then your Nomorobo toll free number is working as intended.

Here's a quick way to test:

a) In your X_InboundCallRoute for Freephoneline, use {ph,ph2,sp1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=3194327596)}

For OBi200 users, use{ph,sp1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=3194327596)}

1866xxxxxxx is your toll free Nomorobo access number. The xxxxxxx needs to be replaced.

b) Using your cellphone, dial your Freephoneline phone number.

The call should be answered within 3 seconds of ringing by Nomorobo ("please try your call again" message is played or something similar), and then call is disconnected by Nomorobo.

c) now, in your X_InboundCallRoute for Freephoneline,
try {ph,ph2,sp1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=yourcellnumber)}

For OBi200 users, try{ph,sp1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=yourcellnumber)}

1866xxxxxxx is your toll free Nomorobo access number. The xxxxxxx needs to be replaced.
Similarly, youcellnumber needs to be replaced with your actual cell phone number.

d) Call your FPL number using your cell phone (or by using another provider other than FPL).

You should see the following in your ATA’s call history (found by logging into ATA with a web browser and navigating to Status–>Call History):
Call Failed (486 Busy Here; SP1(1866xxxxxxx@tollfree.alcazarnetworks.com;ui=yourcellnumber)

That's what's supposed to happen. If there's a busy response from Nomorobo, then the call will continue to ring your phone ports (beyond 3 seconds).

If Nomorobo picks up the call, then your phone ports won't ring.

3194327596 is a known marketer.
http://www.obitalk.com/forum/index.php? ... 6#msg58226
(thanks to azrobert)
There's a related thread here: https://www.obitalk.com/forum/index.php?topic=10368.20.
Should you have further questions concerning Nomorobo, you may want to try asking there. You do not, at this time of this writing, need to setup a dummy SIP trunk as described in the first page of that thread.



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.




How do I tell anonymous callers that I don't accept anon calls with Freephoneline?


So . . .

I saw this thread: https://www.dslreports.com/forum/r31674 ... on-message.

I'm not typically a fan of blocking anonymous callers because some of them can be doctors, hospital admins, etc.
But it's not unreasonable to give important people your smartphone number instead and to block anonymous calls with your FPL number.

For those who remember the steps I took with Tropo before to block telemarketers, the following steps are very similar.

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 this audio file: https://www.dslreports.com/forum/r31680429-
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 anon 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/blockanonmessageaudiofilethatIadded.wav");
say("http://hosting.tropo.com/mytropoaccount#/www/blockanonmessageaudiofilethatIadded.wav");
hangup();
?>
9. Saved the script as anon.php (just has to end with .php)

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

11. Clicked "create app"

12. Stuck {(?|un@@.|Un@@.|anon@.|Anon@.):sp1(mytroponumber)} in X_InboundCallRoute (towards the beginning or left) for FPL in my OBi

(where sp1 is FPL), but it doesn't matter what SP you use, as long as you call your Tropo phone number for free using it.

Rebooted

It does take a good 30 minutes to an hour before new Tropo apps to start working properly with phone numbers.


Anyway, anonymous callers are greeted by out of service SIT tones followed by a message instructing them to reveal their number if they want to reach me (in this example).
I'm not sure if I'm going to use this process for myself yet. But some might be interested.


Good luck!
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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Aug 19, 2003
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Anyone know what the difference between obi202 vs obi212?
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Mar 3, 2002
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Tha DraGun wrote: Anyone know what the difference between obi202 vs obi212?
The OBi212 is the OBi110 replacement.

OBi212 has a line (FXO) port so that you can connect to a PSTN (regular telephone) service or bridge a PSTN service with a VoIP service. OBi212 only has 1 FXS port.
OBi202 has two phone (FXS) ports and no line (FXO) port.

If you want a line port with an OBi202, you need to buy OBiline.
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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Aug 19, 2003
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Webslinger wrote: The OBi212 is the OBi110 replacement.

OBi212 has a line (FXO) port so that you can connect to a PSTN (regular telephone) service or bridge a PSTN service with a VoIP service. OBi212 only has 1 FXS port.
OBi202 has two phone (FXS) ports and no line (FXO) port.

If you want a line port with an OBi202, you need to buy OBiline.
I have a obi110. It's been working great except now every once and a while it will say "line in use" out of the blue. If I don't notice it right away no one can call us. If I unplug the obi and reboot it, it's fine. Until the next time it does the line in use error. So I think I need to finally replace it.

By Connect a regular line (for obi212), you mean connect say a Bell telephone service to the box, to use along side your VoIP? I think I did that before with my obi..
If that's the only difference I Wouldn't need it anymore. so obi202 would be just as good for me.
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Tha DraGun wrote: It's been working great except now every once and a while it will say "line in use" out of the blue.
You might want to read through this thread:
https://www.obitalk.com/forum/index.php?topic=11379.0
By Connect a regular line (for obi212), you mean connect say a Bell telephone service to the box
Yes, that's what the line port is used for on the OBi110 as well.

If you never want to connect Bell (POTS) service, for example, to your ATA, the OBi202 is fine.
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: You might want to read through this thread:
https://www.obitalk.com/forum/index.php?topic=11379.0



Yes, that's what the line port is used for on the OBi110 as well.

If you never want to connect Bell (POTS) service, for example, to your ATA, the OBi202 is fine.
I haven't used a Bell phone line in a long time so I completely forgot that's what the LINE port was for.

I had both ports connected to my house wiring. I must have connected it that way a little while back when I upgraded my router. Since then it had been giving me problems so maybe that was the issue.
I disconnected the LINE port. Hopefully that fixes it and I don't have to buy a new obi
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Tha DraGun wrote: I haven't used a Bell phone line in a long time so I completely forgot that's what the LINE port was for.

I had both ports connected to my house wiring. I must have connected it that way a little while back when I upgraded my router. Since then it had been giving me problems so maybe that was the issue.
I disconnected the LINE port. Hopefully that fixes it and I don't have to buy a new obi
If you're going to connect your house's wiring to the phone port on the ATA, you may need to disconnect your existing telephone company's line at the demarc--or make sure power from it is not running to your existing phone jacks. Otherwise, you run this risk of frying your ATA. Visit http://www.voipmyhouse.com/#thesolution.
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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Mississauga
I hope they have this sale again this year, I need one because my 3 year old one stopped working:(
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djjungly wrote: I hope they have this sale again this year, I need one because my 3 year old one stopped working:(
If I happen to see it on sale, I’ll PM you.
Please do not PM me for tech support. I help out on the forums when I can. Thank you.
OBi200/202 Freephonline PDF guide (version 1.60) can be found here. OBi200 info can be found here. For OBi202 info, click here.

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