Freebies

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

Newbie
Oct 22, 2005
48 posts
1 upvote
Toronto
Webslinger wrote:
Sep 12th, 2016 5:12 am
xxplicit wrote:
Sep 12th, 2016 1:00 am
I'm trying to register for an account at https://www.freephoneline.ca/accountRegistrationStepOne but it is coming back with an error "invalid contact number". I've used both 647 and 416 area code active cell phone numbers with no luck.
Anybody able to register a new account?
I haven't signed up for a new account in quite some time and don't recall a SMS verification step. Must be something new. I'm not going to try to sign up again to see how that works, but it's troubling that they're suddenly limiting accounts to people who have a cellular texting service. I'm not sure what the reason for that would be.

Send a ticket in and ask: http://fongo.help/submit

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

If you receive an answer, please let us know what support says. I would also like to know the answer to your question. Thank you
I tried again just now with my 647 number and was able to successfully register an account. I didn't get an SMS verification, only email. Not sure why it didn't work the past few days but it works now.
Deal Fanatic
User avatar
Mar 3, 2002
9364 posts
3230 upvotes
xxplicit wrote:
Sep 12th, 2016 11:24 am

I tried again just now with my 647 number and was able to successfully register an account. I didn't get an SMS verification, only email. Not sure why it didn't work the past few days but it works now.

Weird. Did you have to use a cell phone number? The reason I'm asking is because the website states, "Contact Number: This number will be used to verify your account by SMS later in the registration process." Seems a bit unfair considering that I suspect some senior citizens do not have cell phones.

Back when I signed up that message was never there.

https://www.freephoneline.ca/accountRegistrationStepOne
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 22, 2005
48 posts
1 upvote
Toronto
Webslinger wrote:
Sep 12th, 2016 11:32 am
xxplicit wrote:
Sep 12th, 2016 11:24 am

I tried again just now with my 647 number and was able to successfully register an account. I didn't get an SMS verification, only email. Not sure why it didn't work the past few days but it works now.

Weird. Did you have to use a cell phone number? The reason I'm asking is because the website states, "Contact Number: This number will be used to verify your account by SMS later in the registration process." Seems a bit unfair considering that I suspect some senior citizens do not have cell phones.

Back when I signed up that message was never there.

https://www.freephoneline.ca/accountRegistrationStepOne
I'm not sure if I had to use a cell but I did. After logging in it needed me to register for the desktop app and that required SMS verification.
Deal Fanatic
User avatar
Mar 3, 2002
9364 posts
3230 upvotes
xxplicit wrote:
Sep 12th, 2016 5:23 pm

I'm not sure if I had to use a cell but I did. After logging in it needed me to register for the desktop app and that required SMS verification.
Thanks for the info!
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.
Deal Guru
User avatar
Jan 27, 2004
10016 posts
1068 upvotes
is there anyway to check how much credit do I have with FPL?
2007 - Ipod Video (TD), Ipod Shuffle (GM)
2006 - Ipod Nano (TD)
2005 - Ipod Shuffle (TD)
Deal Fanatic
User avatar
Mar 3, 2002
9364 posts
3230 upvotes
kiasu wrote:
Sep 12th, 2016 7:41 pm
is there anyway to check how much credit do I have with FPL?
Visit http://forum.fongo.com/viewtopic.php?f= ... 54&p=64214
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.
Deal Addict
Aug 25, 2005
4582 posts
1202 upvotes
Webslinger, any way to block a specific number either with FPL or via the Obihai202 device?

edit: nvm looks like *67######### works
Newbie
Mar 12, 2013
25 posts
4 upvotes
Same question as above, what do I do if I don't have a cell phone for SMS verification?
Deal Fanatic
User avatar
Mar 3, 2002
9364 posts
3230 upvotes
BenK wrote:
Sep 13th, 2016 3:39 pm
Webslinger, any way to block a specific number either with FPL or via the Obihai202 device?
Click newegg-obihai-obi200-ata-49-99-1-50-ehf ... #p26724384

Look at the Telemarketing section.
edit: nvm looks like *67######### works
That doesn't block an incoming call.

That blocks your outgoing caller ID for one call:
http://www.obihai.com/docs/OBiFeatureStarCodes.pdf

I'm not entirely sure what you want, but I covered both possibilities.
Last edited by Webslinger on Sep 13th, 2016 10:50 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.
Deal Fanatic
User avatar
Mar 3, 2002
9364 posts
3230 upvotes
billiondollarman wrote:
Sep 13th, 2016 7:36 pm
Same question as above, what do I do if I don't have a cell phone for SMS verification?

Send a ticket in and ask: http://fongo.help/submit

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

If you receive an answer, please let us know what support says. I would also like to know the answer to your question. 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.
Deal Addict
Aug 25, 2005
4582 posts
1202 upvotes
Webslinger wrote:
Sep 13th, 2016 8:20 pm
BenK wrote:
Sep 13th, 2016 3:39 pm
Webslinger, any way to block a specific number either with FPL or via the Obihai202 device?
Click newegg-obihai-obi200-ata-49-99-1-50-ehf ... #p26724384

Look at the Telemarketing section.
edit: nvm looks like *67######### works
That doesn't block an incoming call.

That blocks your outgoing caller ID for one call:
http://www.obihai.com/docs/OBiFeatureStarCodes.pdf

I'm not entirely sure what you want, but I covered both possibilities.
thanks, i was looking to block a specific number from calling us. some crazy telemarketer who won't give up. Has called more than 10 times since Sept 1 according to FPL log.
Deal Fanatic
User avatar
Mar 3, 2002
9364 posts
3230 upvotes
BenK wrote:
Sep 14th, 2016 11:37 am

thanks, i was looking to block a specific number from calling us. some crazy telemarketer who won't give up. Has called more than 10 times since Sept 1 according to FPL log.
Okay, yeah, I'll post this here again:

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 (This doesn't apply to Freephonline; I'm just using this as an example)

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

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 #
@@. = any length alphanumeric sequence except #

un@@. will catch unknown
anon@. will catch anonymous

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

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.
Deal Fanatic
User avatar
Mar 3, 2002
9364 posts
3230 upvotes
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. For Bell Hubs, visit please-sticky-how-bypass-bell-hub-use-y ... r-1993629/


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-obihai-obi202-ata-79-99-1-50-ehf ... #p26724384

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

Image

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 after registering and logging into the forums: 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.
Last edited by Webslinger on Sep 16th, 2016 1:37 pm, 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.
Deal Fanatic
User avatar
Mar 3, 2002
9364 posts
3230 upvotes
Getting one way audio issues with an SPA122 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)?

Try a different phone; check all cables and cords.

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.

Hardware Specific Issues

A. Netgear R7000 routers

Update firmware. 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 third party XWRT-Vortex firmware. I recommend doing this anyway to obtain easy access to both UDP Unreplied and UDP Assured timeout settings. 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. In your router, navigate to Advanced Settings–>WAN–>NAT Passthrough–>SIP Passthrough. Change SIP Passthrough to “Enabled + NAT helper.” Click “Apply.”

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. Asus VLAN

A number of people have been trying to eliminate Bell Hubs from their setups by using this guide: http://blog.ngpixel.com/post/1044497475 ... own-router.
At the time of this guide being written, NAT acceleration must be disabled in this setup in order for SIP services, including Freephoneline, to work properly. In your router, navigate to Advanced Settings-->LAN-->Switch Control-->NAT Acceleration. Select "disable." Click "apply."Then reboot your modem, router (wait for Wi-Fi SSIDs to populate first before rebooting ATA), and your ATA, in that order.

To determine whether you need NAT Acceleration enabled, visit https://routerguide.net/nat-acceleration-on-or-off/. If you do require NAT Acceleration to be enabled, don’t use VLAN with Asus routers.

D. Hitron CGN series gateway modem/router combos (from Rogers, Shaw, or another ISP) or any modem/router combo from any ISP with SIP ALG forced on

If you don’t have your own router, and if you can’t get someone from Rogers or your ISP to disable SIP ALG for you in their modem/router combo, your ATA may need to register with voip4.freephoneline.ca:6060. The purpose of voip4.freephoneline.ca:6060 is to help circumvent faulty SIP ALG features in routers. So, if you’re experiencing one-way audio issues as a result of SIP ALG, this is the SIP server to try. Check to ensure that you can’t disable SIP ALG yourself (refer to point E below).

E. Hitron CGN3ACSMR and CODA-4582 series gateway modem/router combos from Rogers (and possibly other ISPs)
Open your web browser, and login at 192.168.0.1. Default username is cusadmin.
Select the “Basic” tab and disable “SIP ALG.” Click the “save changes” button.







Before you begin,

i)disable DMZ and all port forwarding in your router,

and

ii) 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. For Bell Hubs, visit please-sticky-how-bypass-bell-hub-use-y ... r-1993629/



1) Make sure your ringer is enabled on your phone. Try a different phone; check all cables and cords.

2) Log into your SPA122

3) If you have your own router or are using a modem/router combo that's not in bridge mode and are connecting the SPA122 to the router or modem/router combo, stick your SPA122 in bridge mode:

a. Navigate to Network Setup menu tab-->Network Service menu (on the left)
b. Select Network Service drop-down box
c. Select Bridge
d. click Submit

The last thing you need with FPL is double NAT issues.


4.

A. Under Voice-->User 1 and User 2 (both of these)-->navigate to Supplementary Service Settings

Ensure

a) DND setting is set to NO
b) Block ANC Setting is set to NO
c) DND Activated is set to NO
d) Cfwd All Serv: no
e) Cfwd Busy Serv: no
f) Cfwd No Ans Serv: no


B. Under Voice-->User 1 and User 2 (both of these)
http://sbkb.cisco.com/CiscoSB/GetArticl ... onverted=0

Check Call Forward Settings


a) Cfwd All Dest:

That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.

b) Cfwd Busy Dest

That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.


c) Cfwd No Ans Dest

That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.


5. In your SPA122, Navigate to Voice-->Line 1 (or whatever you're using for FPL)-->SIP settings, change SIP Port to a random number between 30000 and 60000.


6. In your SPA122, Navigate to the Voice-->SIP-->NAT Support Parameters, and make sure that the following settings are enabled:

a)Handle VIA received-->yes
b)Handle VIA rport-->yes
c)Substitute VIA Addr-->yes

save settings

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 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.

If you're using a modem/router combo issued by your ISP, contact your ISP. Be aware that most reps will have no clue what SIP ALG is, much less know how to disable it.
You may need to ask to speak to a tier 2 rep. SIP ALG features are sometimes hidden or blocked off to customers in modem/router combos issued by ISPs.

To understand why SIP ALG is often buggy in routers and causes horrible problems, visit https://www.voip-info.org/wiki/view/Routers+SIP+ALG.

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

For reference, this is how to disable SIP ALG in Asus routers, although I've never had to do this for Freephoneline: https://kb.intermedia.net/article/3024

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

Image

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.

9. Try Proxyserver voip4.freephoneline.ca:6060
http://forum.fongo.com/download/file.php?id=1742 (look at page 4 of the .pdf)


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).



10. Try this at your own risk: use voip3.freephoneline.ca as the proxyserver

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 #12.

11. 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.

12. If none of that helps, then, unfortunately, you're pretty much stuck with port forwarding your RTP (UDP) port range 16384-16482 from your router to your ATA. For reference, that range can be found under Voice-->SIP-->RTP Parameters-->RTP Port Min and RTP Port Max. You're going to want to double check those numbers in your ATA. 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.

13. 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.

14. Thanks to Mango, many of us now understand that in order for ATAs to remain registered and working properly
with a VoIP SIP provider like Freephoneline, in particular after power failures, the following conditions must be met:

UDP Unreplied Timeout (in your router) < NAT Keep-alive Interval < UDP Assured Timeout (in your router) < SIP Registration Failure Retry Wait Time (Reg Retry Intvl)

“<“ means less than.

A problem can occur when the Keep-Alive interval is greater than UDP Assured Timeout (often 180 by default in
consumer routers): the NAT hole will close due to the ATA not communicating frequently enough with the SIP server.
In turn, incoming calls may, intermittently, not reach the ATA. Again, NAT Keep-alive Interval is supposed to be 20 with
FPL.

Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers may be
difficult, if not impossible. Asuswrt-Merlin, third party firmware for Asus routers, does offer easy access to these two
settings, which are found under General–>Tools-->Other settings. In part, for this reason, I tend to use Asus routers
that work with Asuswrt-Merlin. However, my understanding is that third party Tomato firmware has these two settings
as well. So if your router supports Tomato firmware, that may be another option.
The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time (Reg Retry Intvl setting in your ATA) is 120. I use 10 for UDP
Unreplied Timeout and 117 for UDP Assured Timeout.

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.
When creating a ticket, for the issue type select VoIP Unlock Key-->My account inquiry. Ask for a "forced registration."
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.
Last edited by Webslinger on Sep 16th, 2016 12:55 pm, 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.
Deal Fanatic
User avatar
Mar 3, 2002
9364 posts
3230 upvotes
Getting one way audio issues with a PAP2T-NA 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.

Preliminary Steps

i) Make sure your ringer is enabled on your phone. Check to ensure all cords are properly seated and working properly. Try a different phone.

ii) Log into your PAP2T

A. Under User 1 and User 2 tabs (both of these)

B. Select Advanced View

C. Under Supplementary Service Settings

Ensure

a) DND setting is set to NO
b) Block ANC Setting is set to NO
c) DND Activated is set to NO

D. Under User 1 and User 2 tabs (both of these)

Check Call Forward Settings

a) Cfwd All Dest:

That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.

b) Cfwd Busy Dest

That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.


c) Cfwd No Ans Dest

That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.

E. Ensure that you're using the correct SIP Username and Password: login at https://www.freephoneline.ca/showSipSettings.



Hardware Specific Issues

A. Netgear R7000 routers

Update firmware. 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 third party XWRT-Vortex firmware. I recommend doing this anyway to obtain easy access to both UDP Unreplied and UDP Assured timeout settings. 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. In your router, navigate to Advanced Settings–>WAN–>NAT Passthrough–>SIP Passthrough. Change SIP Passthrough to “Enabled + NAT helper.” Click “Apply.”

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. Asus VLAN

A number of people have been trying to eliminate Bell Hubs from their setups by using this guide: http://blog.ngpixel.com/post/1044497475 ... own-router.
At the time of this guide being written, NAT acceleration must be disabled in this setup in order for SIP services, including Freephoneline, to work properly. In your router, navigate to Advanced Settings-->LAN-->Switch Control-->NAT Acceleration. Select "disable." Click "apply."Then reboot your modem, router (wait for Wi-Fi SSIDs to populate first before rebooting ATA), and your ATA, in that order.

To determine whether you need NAT Acceleration enabled, visit https://routerguide.net/nat-acceleration-on-or-off/. If you do require NAT Acceleration to be enabled, don’t use VLAN with Asus routers.

D. Hitron CGN series gateway modem/router combos (from Rogers, Shaw, or another ISP) or any modem/router combo from any ISP with SIP ALG forced on

If you don’t have your own router, and if you can’t get someone from Rogers or your ISP to disable SIP ALG for you in their modem/router combo, your ATA may need to register with voip4.freephoneline.ca:6060. The purpose of voip4.freephoneline.ca:6060 is to help circumvent faulty SIP ALG features in routers. So, if you’re experiencing one-way audio issues as a result of SIP ALG, this is the SIP server to try. Check to ensure that you can’t disable SIP ALG yourself (refer to point E below).

E. Hitron CGN3ACSMR and CODA-4582 series gateway modem/router combos from Rogers (and possibly other ISPs)
Open your web browser, and login at 192.168.0.1. Default username is cusadmin.
Select the “Basic” tab and disable “SIP ALG.” Click the “save changes” button.





In addition to what's written below, enable QoS properly in your router for your ATA.
https://www.obitalk.com/info/faq/Troubl ... ing-choppy (not just for choppy audio; if all your bandwidth is being eaten up by other devices using the internet, calls won't occur either)


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."




Full Steps

Do the following, slowly and carefully, all the way down until the problem is resolved:

1. 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. For Bell Hubs, visit please-sticky-how-bypass-bell-hub-use-y ... r-1993629/

2. Disable DMZ and all port forwarding in your router. Port forwarding is a security risk.


3. In your PAP2T, Navigate to Line 1 (or whatever you're using for FPL)-->SIP settings, change SIP Port to a random number between 30000 and 60000


4. In your PAP2T, Navigate to the SIP tab-->NAT Support Parameters, and make sure that the following settings are enabled:

a)Handle VIA received-->yes
b)Handle VIA rport-->yes
c)Substitute VIA Addr-->yes

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 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

Image

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.

7. Try Proxyserver voip4.freephoneline.ca:6060
visit http://forum.fongo.com/viewtopic.php?f=15&t=16294 (look at page 6 of the .pdf)


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).



8. Try this at your own risk: use voip3.freephoneline.ca as the proxyserver

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. For the issue type, select VoIP Unlock Key–>My Account Inquiry. Ask for a “forced account registration.”

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 #10.

9. 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.

10. If none of that helps, then, unfortunately, you're pretty much stuck with port forwarding your RTP (UDP) port range 16384-16482 from your router to your ATA. For reference, that range can be found under SIP-->RTP Parameters-->RTP Port Min and RTP Port Max. You're going to want to double check those numbers in your ATA. 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.

11. 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.


12. Thanks to Mango, many of us now understand that in order for ATAs to remain registered and working properly
with a VoIP SIP provider like Freephoneline, in particular after power failures, the following conditions must be met:

UDP Unreplied Timeout (in your router) < NAT Keep-alive Interval < UDP Assured Timeout (in your router) < SIP Registration Failure Retry Wait Time (Reg Retry Intvl)

“<“ means less than.

A problem can occur when the Keep-Alive interval is greater than UDP Assured Timeout (often 180 by default in
consumer routers): the NAT hole will close due to the ATA not communicating frequently enough with the SIP server.
In turn, incoming calls may, intermittently, not reach the ATA. Again, NAT Keep-alive Interval is supposed to be 20 with
FPL.

Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers may be
difficult, if not impossible. Asuswrt-Merlin, third party firmware for Asus routers, does offer easy access to these two
settings, which are found under General–>Tools-->Other settings. In part, for this reason, I tend to use Asus routers
that work with Asuswrt-Merlin. However, my understanding is that third party Tomato firmware has these two settings
as well. So if your router supports Tomato firmware, that may be another option.
The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time (Reg Retry Intvl setting in your ATA) is 120. I use 10 for UDP
Unreplied Timeout and 117 for UDP Assured Timeout.


13. 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. For the issue type, select VoIP Unlock Key–>My Account Inquiry. In addition to explaining your issue, request a “forced registration.”
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.


Proper device reboot order is always modem--->router--->ATA in that order. Wait for one to be fully up and running before turning on or rebooting the next device.
Last edited by Webslinger on Sep 16th, 2016 1:31 pm, 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.

Top

Thread Information

There is currently 1 user viewing this thread. (1 member and 0 guests)

link_1983