[Resolved] can't receive calls from wife's workplace

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[Resolved] can't receive calls from wife's workplace

Postby Fixmyphone » 09/28/2022

I am receiving calls from most numbers but not all.

I have checked my call logs on FreePhoneLine.ca and the calls are listed under the incoming but they go straight to my voicemail or to dead air for the caller. This has been going on for a couple months but was never an issue previously.

I have gone through all the setting again today and everything is set up properly on my end.

The numbers that are not working are originating from my wifes work so they would be behind an extension. I have her checking to see who their provider is and if it has changed recently as this never use to be an issue.

I am not sure if there is any other steps I can take my end as I have updated my OBI202's firmware, checked all the setting, ensured that the SIP setting is correct on my ASUS router, etc. but still just calls from her work are not making it through even though they appear on my call logs on FreePhoneLine.

Any suggestions would be great.
Fixmyphone
Just Passing Thru
 
Posts: 3
Joined: 09/28/2022
SIP Device Name: OBI202
Firmware Version: 3.2.2 (Build: 8680EX
ISP Name: Rogers Ignite
Computer OS: Windows 10 X64
Router: ASUS RT-AX86U

Re: Receiving calls

Postby Liptonbrisk » 09/28/2022

Fixmyphone wrote:I have gone through all the setting again today and everything is set up properly on my end.


The PDF guide you should be using is located at the bottom of the first post from viewtopic.php?f=15&t=18805#p73839.
Using Obitalk's default FPL profile is undesirable, as explained on pages 6 and 7.

Please use the PDF guide fully to setup your OBi202 and when checking settings.

ensured that the SIP setting is correct on my ASUS router



1. Enable bridge mode in whatever gateway Rogers gave you: https://www.rogers.com/support/internet ... your-modem.


2a) Disable SIP Passthrough (SIP ALG) when using stock (official) Asus router firmware. Login to your router's web UI. Navigate to Advanced
Settings–>WAN–>NAT Passthrough
Disable SIP Passthrough. With SIP Passthrough enabled, incoming calls from Rogers/Fido numbers won’t work.

b) For Asuswrt-Merlin (currently, version 386.7_2 at the time of writing), I use “Enabled” for SIP Passthrough.
With Merlin, ensure that you are not using “+NAT helper”, which is the ALG. With SIP Passthrough set to "Enabled + NAT Helper", incoming calls from Rogers/Fido numbers won’t work
.

I recommend using Asuswrt-Merlin instead of stock Asus firmware: https://www.asuswrt-merlin.net/about.
(However, I will not be held responsible for problems arising from failed firmware updates or user error.)

See point D from viewtopic.php?f=8&t=20199#p78976, and adjust "UDP Timeout: Assured" and "UDP Timeout: Unreplied" in Asuswrt-Merlin web UI (login to router; navigate to General-->Tools-->Other), respectively. You can't do that using stock (official) Asus router firmware.


3. If you use the Obitalk web portal (http://www.obitalk.com) to configure your ATA, keep in mind
that you must continue using it to configure your ATA unless you disable Obitalk
Provisioning first. Otherwise whatever settings you change will eventually be overwritten
(they will be transferred from your Obitalk.com account to your ATA) by what you
previously entered at obitalk.com anyway. If you wish to disable this behaviour, dial ***1.
Enter the IP address you hear into a web browser. Navigate to System
Management-->OBiTalk Provisioning-->select Disabled for the method. Save. Reboot ATA.
Afterwards, 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.

If you use Obitalk.com, refer to pages 10 and 11 of the PDF guide first to learn how to enter the "Expert" menus.

4. Dial ***1.
Enter the IP address you hear into a web browser.

a) In your Obihai ATA or at Obitalk.com (whichever method you use), navigate to Voice Services-->SP(FPL) Service-->X_UserAgentPort
X_UserAgentPort should be a random port number between 30000 and 60000. Just pick a port number in that range.
If you already have a random number in that range, simply choose a new one.

By using a high random port you help to thwart SIP scanners/hackers.

b) Navigate to Service Providers-->ITSP Profile (FPL)-->SIP
i) ensure X_DiscoverPublicAddress is enabled (it is by default)

This helps to ensure the ATA has discovered your public WAN IP.

c) enable X_UsePublicAddressInVia (it's not by default)
You will need to uncheck default, device default, and Obitalk settings boxes (if you're using Obitalk). Then check the box to enable the
feature. Submit changes.

This helps to ensure data from FPL is sent to your public WAN IP address instead of the ATA's LAN IP (or outer space. That is, FPL sending packets to 10.0.0.111, for example, which is a LAN IP, isn't going to accomplish much).


d) Navigate to Voice Services-->SP(FPL)
i) X_KeepAliveEnable should be Checked
ii) X_KeepAliveExpires should be 20 seconds
iii) X_KeepAliveMsgType should be "notify"

Submit changes.

This helps to keep NAT associations/connections working (alive).

e) Navigate to Service Providers–>ITSP Profile (FPL)–>SIP
i) RegistrationPeriod should be 3600 seconds

If it's not, you run the risk of being temporarily IP banned by FPL's server.

ii) RegisterRetryInterval should be at least 120 seconds

f) Submit changes.

g) Reboot Rogers gateway (wait for it to be fully up and running)-->Reboot Asus router (wait for it to be fully up and running first, including broadcasting Wi-FI SSIDs)-->then finally reboot OBi202

h) Test incoming calls.

I have checked my call logs on FreePhoneLine.ca and the calls are listed under the incoming but they go straight to my voicemail or to dead air for the caller


Oh, the calls are actually reaching Fibernetics' network. If your wife is using another SIP service for outbound calls, the issue could be on her end.


The calls aren't listed in your ATA's call history, correct?

To check, dial ***1.
Enter the IP address you hear into a web browser.
Navigate to Status-->System Status-->Call History

The next time your wife calls from work again, during the active call, navigate to status-->call status-->
Check Packets Lost, Packet Loss Rate, and Packet Drop Rate. Those should be 0.
Usually, pressing F5 on a keyboard will refresh or update current stats while on the call.

Image

MOS (mean opinion score) score represents call quality. 4.4 is supposed to be the highest score possible when using the G.711u audio codec.



I have her checking to see who their provider is


Visit https://freecarrierlookup.com/, and enter the main business number (without the extension). Answer the question, and click "submit".
Who is the carrier, and do the results indicate it's a wireless carrier?

Well, I doubt she's using another Fongo service. Otherwise, I would have asked to check to ensure X_AcceptSipFromRegistrarOnly is disabled in your ATA as well (Voice Services-->SP Service--> X_AcceptSipFromRegistrarOnly).


It might be worthwhile to see whether you can reproduce this issue using Fongo Mobile using cellular data only: https://www.fongo.com/services/fongo-mobile/.
If incoming calls don't work while using cellular data, submit a ticket as a Fongo Mobile customer: https://support.fongo.com/hc/en-us/requests/new.
Provide them with your wife's number and also the corresponding call log entries from https://account.fongo.com/admin/reports/call-logs/.
However, I would probably also start considering whether the issue is with the phone service at your wife's workplace in that case.

If Fongo Mobile works for incoming calls from your wife's workplace while using cellular data, try again while your smartphone is connected to Wi-Fi.
If incoming calls don't work while connected to Wi-Fi at the point, the problem involves your LAN (or possibly Rogers if the problem involves packet loss or prime time/local node congestion in your area).
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk
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Posts: 2763
Joined: 04/26/2010
SIP Device Name: Obihai 202/2182, Groundwire
Firmware Version: various
ISP Name: FTTH
Computer OS: Windows 64 bit
Router: Asuswrt-Merlin & others

Re: can't receive calls from wife's workplace only

Postby Liptonbrisk » 09/28/2022

Another possibility is an audio codec mismatch.
If it is, then I suspect an issue on the workplace's end, if all other incoming calls work.

FPL only supports G.711u and G.729a.

Are you able to call her workplace using FPL?
If you are then an audio codec mismatch seems unlikely.

First go through the rest of the steps I listed (step by step, down the list). Getting your Rogers Ignite gateway in bridge mode and changing the NAT Passthrough setting correctly in your Asus router should be the first steps taken, especially if your wife's workplace is using Rogers, Fido, Chatr, Cityfone, Zoomer, etc. ATA should be connected to the Asus router, of course.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk
Technical Support
 
Posts: 2763
Joined: 04/26/2010
SIP Device Name: Obihai 202/2182, Groundwire
Firmware Version: various
ISP Name: FTTH
Computer OS: Windows 64 bit
Router: Asuswrt-Merlin & others

Re: can't receive calls from wife's workplace only

Postby Fixmyphone » 09/29/2022

Thanks for trying to help.

I can call my wife's work without issue and I am able to receive all calls from 800 numbers, anonymous numbers, cells, landlines, etc. that I am aware of.

It is only when she calls out from work to our home. This issue 100% originates from just her work and just calls from her work show up in the call logs at freephoneline as 1:00 minute calls but do not show in the call logs on my OBI so something between freephoneline and my OBI doesn't work for just calls originating from her work. Not sure if it is related but she was always able to call before, but as of September 7th, her calls stopped coming through. I have also had to restart my OBI several times around this date for about a week as calls were not connecting from freephoneline for some reason but has not issues since...I am not sure if something was going on but saw other reports of issues in the beginning of September with freephoneline.

I wouldn't put it past my wife's work, municipal town job, to have some half set up corporate VOIP being mishandled by their IT department. My wife is checking to see if they are using some VOIP system through Rogers or similar provider.
Fixmyphone
Just Passing Thru
 
Posts: 3
Joined: 09/28/2022
SIP Device Name: OBI202
Firmware Version: 3.2.2 (Build: 8680EX
ISP Name: Rogers Ignite
Computer OS: Windows 10 X64
Router: ASUS RT-AX86U

Re: can't receive calls from wife's workplace only

Postby Liptonbrisk » 09/29/2022

Fixmyphone wrote:Thanks for trying to help.


Can you at least tell me whether your Rogers modem/router combo is in bridge mode and whether you have SIP Passthrough in your Asus router set as I've described?

What is your router's firmware version? Are you using Merlin? Do you have UDP timeouts set properly?

I wouldn't put it past my wife's work, municipal town job, to have some half set up corporate VOIP being mishandled by their IT department. My wife is checking to see if they are using some VOIP system through Rogers or similar provider.


Are you unable to enter the main business number at https://freecarrierlookup.com/? Enter the main business number (without the extension). Answer the question on the website, and click "submit".
Who is the carrier, and do the results indicate it's a wireless carrier (probably not)?

If it's a toll-free number, try here instead: https://www.800forall.com/SearchWhoOwns.aspx.


This issue 100% originates from just her work and just calls from her work show up in the call logs at freephoneline as 1:00 minute calls



Freephoneline call logs always round to the nearest minute now, regardless of call duration: viewtopic.php?f=8&t=20404#p79878.
What is the listed "disconnect reason"?

but do not show in the call logs on my OBI


Okay, I was wondering whether OBi202 was dropping the call, but it's not even reaching your ATA.

so something between freephoneline and my OBI doesn't work for just calls originating from her work.


Yes, it could be a SIP ALG issue if you didn't stick the Rogers device in bridge mode and if you still have SIP Passthrough enabled in stock (official) Asus firmware.
viewtopic.php?f=8&t=20182#p78916

Your ATA could also be, randomly, losing connectivity if your internet service is unreliable.

The instructions below are for testing to FPL's proxy servers, but I would also test to 208.85.218.149 and 208.85.218.150, which are the RTP IPs at the moment for FPL.
That's where the RTP audio packets (audio stream) come from.

"Test pings and jitter (you want little to no variation between pings) to the specific Freephoneline SIP servers you plan on using.

Use winmtr: https://sourceforge.net/projects/winmtr/. Ping about 200 times to each server.

My pings to
-voip.freephoneline.ca average 11 ms.
-voip2.freephoneline.ca average 12 ms
-voip4.freephoneline.ca average 27 ms

If you're using a Macintosh, maybe this helps: https://www.reddit.com/r/TagPro/comment ... tr_on_mac/

When using WinMTR, look at the very last hop or line. Look at your average ping and then maximum ping. Although WINMTR doesn't provide a jitter value, you can get an idea of what yours is by subtracting maximum ping from your average. Jitter is the difference between each successive ping. The bigger the difference, the bigger the problem.

Same with ping, which represents lag or delay. The lower your ping and jitter, the better.

You do not want high pings and lots of jitter (you do not want a lot of variation between each ping). If you get horrible results (pings over 200ms), to any server, you probably don’t want to use that server. So you would want to give that server the lowest priority.

I get between 11 (voip.freephoneline.ca and voip2.freephoneline.ca) and 24ms (voip4.freephonline.ca) on average, depending on the server I'm testing to. Preferably, you want pings below 100ms.
Anything over 200ms is unacceptable.
What you don't want to see is 40, 45, 50, 35, 500, 40, 30, 45, 700. That's bad jitter.
You want relatively consistent pings without a lot of variation.

One reason why jitter can occur is due to other devices on your LAN (local area network) using bandwidth. That’s why properly enabling QoS in your router for your ATA is always a good idea. Refer to point C from viewtopic.php?f=8&t=20199#p78976.

Bad jitter can produce broken-up audio or choppiness during phone calls. Severe jitter (or large ping spikes) can cause calls to drop (and incoming calls won’t arrive while the ping spike is occurring). Ping affects delay.

I recommend testing pings/jitter between 8 p.m. and 11 p.m. to see if local congestion is a factor (this often is your ISP's fault). Sundays are the best days to test (because that's when most people in your area will be home). 8 p.m. - 11 p.m. is prime time. During prime time (between 8 p.m. and 11 p.m.) cable internet nodes may be oversubscribed in your area and face congestion issues (and congestion can also exist with DSL). So I suggest testing services between 8 p.m. and 11 p.m., particularly on Sundays, when everyone in your area will be home.

Ping is a measurement of data packet transmission, and ping does affect delay or lag. All gamers know, almost inherently, that lag affects them negatively. A PC gamer will pound his or her keyboard in hope that a character will respond on his or her monitor, quickly, but when there's a delay or lag, reality doesn't meet expectation. A gamer can see this problem visually. Over VoIP, anything over 200-210 ms, you will typically start to encounter crosstalk due to increased delay, even if the untrained ear doesn't notice. All VoIP services are subject to the same scientific principles including the fact that speed of transmission affects delay, and Freephoneline is not some magical service that is somehow exempt from issues arising from high pings and jitter. When pings and, especially, jitter are high, it's a pretty horrible experience, just as it would be with any other VoIP service. When pings and jitter are fine, Freephoneline is great.

Lastly, anyone using any communication service (or even when playing online games or using other online services) should understand that the longer the path to the server being used, the greater the potential exists for a problem to occur somewhere along that path. Freephoneline’s SIP servers are located in Ontario."

-- from download/file.php?id=2195 (pages 16 and 17)

Also, please note that only one registration per FPL account is allowed at any time. When there are multiple devices/softphones using the same account, only the most recent registration is valid. The previous device will lose registration (despite what the ATA's registration status indicates, since 3600 seconds is a long time to update registration status), and incoming calls will not work on it. Registration is required for incoming calls but not for outgoing calls.This is especially important to consider if someone else is using your SIP credentials (username and password) that are found after logging in at https://www.freephoneline.ca/showSipSettings (or if you're trying to register your FPL account with a smartphone SIP app, with another device, or, in this case with an additional SP on your OBi202). Registration is required for incoming calls. This is also important to consider if you're using Freephoneline's desktop application (don't have it running while using your ATA with the same FPL account). Additionally, keep in mind that if someone else is also attempting to register the same SIP credentials on another device where you live, too many registration attempts can result in a temporary IP ban. If you ever see a SIP user agent that you don't recognize after logging in at the above link, someone else is using your credentials (possibly, you've been hacked in that scenario).




One of my routers is the same one that you're using, except I use Asuswrt-Merlin. I have UDP timeouts set properly.
My ISP's gateway/hub is being used in bridge mode (so I don't have to worry about its router features interfering with anything).
Obviously, I'm not experiencing incoming call issues from anyone that I know (Telus, Rogers, Bell, Freedom, etc.), or I'd be posting about it.
The SIP service customers that I know are all able to call me without any issue, although I can't say that I know many.

My OBi202 is configured exactly as outlined in this PDF guide: download/file.php?id=2195.


Not sure if it is related but she was always able to call before, but as of September 7th, her calls stopped coming through.


Something probably changed. There's a number of possibilities including intermittent problems with your internet service.

The call is reaching Fibernetics. It's in Freephoneline's call logs for your account. Unless your ATA is losing registration, a UDP timeout issue is causing intermittent problems, your Rogers internet service is unreliable at your location, there's a mangled SIP header involved, or something weird going on with the switches FPL uses, I'm not sure why the call wouldn't reach your router. That doesn't make sense if your ATA is registered, if SIP Passthrough is set properly in your router with bridge mode enabled in your Rogers modem/router combo, and if UDP timeouts are set properly in your Asus router. Make sure to enable X_UsePublicAddressInVia in your ATA.

I have also had to restart my OBI several times around this date for about a week as calls were not connecting from freephoneline


voip.freephoneline.ca wasn't working for incoming calls on September 15th for some accounts. Otherwise, there weren't reported issues.

My first inclination is to suspect a problem with Rogers in your area (which you can try to confirm by following the WinMTR instructions in the PDF guide when a problem occurs), unless, of course, you haven't switched to Asuswrt-Merlin and set UDP timeouts properly.


I wouldn't put it past my wife's work, municipal town job, to have some half set up corporate VOIP being mishandled by their IT department


If that's the case, it seems strange that the outgoing call is directed all the way to your FPL account but not to your ATA.
If they're using a SIP service provider on their side, I wonder whether SIP ALG is enabled on their end.


I was thinking of asking you to collect a syslog from the OBi202, but that’s not useful since the call isn't reaching your ATA anyway.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk
Technical Support
 
Posts: 2763
Joined: 04/26/2010
SIP Device Name: Obihai 202/2182, Groundwire
Firmware Version: various
ISP Name: FTTH
Computer OS: Windows 64 bit
Router: Asuswrt-Merlin & others

Re: can't receive calls from wife's workplace only

Postby Liptonbrisk » 09/29/2022

The call may not even be reaching your router, particularly if the call is dropping immediately to FPL's voicemail system, but in order to check you would need to examine its system log: https://www.asus.com/support/FAQ/1044954/. You would need to enable logging of dropped packets (Advanced Settings->Firewall-->General), and then check for them in conjunction with your ATA's LAN IP in your Asus router's syslog to determine whether your router is blocking the incoming call. If you need help with this, I suggest asking at https://www.snbforums.com/ in the appropriate forum (my time here is limited as it is). However, if you do have everything configured according to the PDF guide I linked previously, have enabled bridge mode in your Rogers gateway, have NAT Passthrough (that is a big deal) set properly, and have UDP timeouts set properly (can also be a big deal), then I am extremely skeptical that your router is is responsible for blocking the call unless you've made other changes.

The easiest way for you to check whether your Asus router is blocking anything is to remove it. Enable bridge mode in your Rogers modem/router combo: https://www.rogers.com/support/internet ... your-modem. Attach the OBi202 to the back of your Rogers modem/router combo. Disconnect all other devices. Keep in mind that any device connected to your Rogers modem/router combo (gateway) will not be protected by it while it's in bridge mode (firewall in Rogers gateway is dropped). So only do this temporarily while testing incoming calls from your wife's workplace.


The situation is odd. The reasons why (that I can think of at the moment) for why the call isn't reaching the ATA after reaching FPL's network include your ATA periodically losing registration, experiencing packet loss (intermittent issues) with Rogers at the time of the call, FPL sending data to something other than your public WAN IP (make sure to enable X_UsePublicAddressInVia in your ATA) , a UDP timeout issue in your router (use Asuswrt-Merlin and adjust UDP timeouts as mentioned earlier) that is not resetting properly but eventually does so often enough for you to not notice a problem exists, or some other issue involving your router (make sure SIP Passthrough is set properly).

You don't have Follow Me enabled after logging in at https://www.freephoneline.ca/followMeSettings, correct? If so, disable it while testing.


Anyway, if you can reproduce the issue while using Fongo Mobile, when your smartphone is connected to cellular data (https://www.fongo.com/services/fongo-mobile/), submit a ticket and provide the appropriate call history log entries, which you can find after logging in at https://account.fongo.com/admin/reports/call-logs/). You can submit a technical support request ticket as a Fongo Mobile user: https://support.fongo.com/hc/en-us/requests/new. Fongo Mobile is free to use. Cellular data charges depend on your smartphone service plan/agreement. If the incoming call doesn't work while the smartphone is connected to your Asus router's Wi-Fi (but does work while using cellular data instead), then I suspect you need to look more closely at your LAN (Asus router) and OBi202 configuration and possibly Rogers' (intermittent modem signal/noise problems, packet loss) internet service at your location.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk
Technical Support
 
Posts: 2763
Joined: 04/26/2010
SIP Device Name: Obihai 202/2182, Groundwire
Firmware Version: various
ISP Name: FTTH
Computer OS: Windows 64 bit
Router: Asuswrt-Merlin & others

Re: can't receive calls from wife's workplace only

Postby Fixmyphone » 10/06/2022

First off thank you for the replies.

It seems that it wasn't just my wife's work as I started having multiple calls to straight to voicemail and never making it to my ATA.

I connected my OBI directly to my Rogers modem and it worked so I knew it was my Router causing the issues.

I went through all the settings on my OBI and my Asus router to make sure everything was correct.

I manually checked for any firmware updates for my router and there was one that was a few days old.

Saved my settings, update my router, reset my router post update and uploaded my saved settings.

While double checking that my settings were good my wife called from work with no issues.

So it was the Asus update that I ran sometime in August and never knew it was an issue as my wife will usually call from her cell while going for a walk on lunch and just recently has been calling from her work phone to check in on the kids when they come home for lunch.

So end of day it was the Asus firmware which everything works after the update I just installed.

I will look into switching to Merlin for my router as it seems to be more solid once properly configured.

Thank you again for all your help and if I have problems in the future I now know how to trouble shoot my way through most of them.

Thanks!!!
Fixmyphone
Just Passing Thru
 
Posts: 3
Joined: 09/28/2022
SIP Device Name: OBI202
Firmware Version: 3.2.2 (Build: 8680EX
ISP Name: Rogers Ignite
Computer OS: Windows 10 X64
Router: ASUS RT-AX86U

Re: can't receive calls from wife's workplace only

Postby Liptonbrisk » 10/06/2022

Fixmyphone wrote:
I went through all the settings on my OBI and my Asus router to make sure everything was correct.


Okay, ensure SIP Passthrough is disabled when using stock (official) Asus firmware.

So end of day it was the Asus firmware which everything works after the update I just installed.


I've only tested (briefly) with the latest stock Asus firmware version, which did work. Otherwise, I haven't used stock Asus firmware in ages.


I will look into switching to Merlin for my router as it seems to be more solid once properly configured.


Yes, and according to the developer, using Asuswrt-Merlin doesn't void your router's warranty: https://www.snbforums.com/threads/warra ... ost-636124.

Run your Rogers gateway in bridge mode, and use Merlin. Ensure SIP Passthrough does not show "+ NAT Helper". I have SIP Passthrough set to "Enabled" with Merlin (but not "+NAT Helper).

"+ NAT Helper" is the ALG in Merlin.

I don't necessarily advise doing this (it works though, at least when using Merlin), but if you want to keep your router's UDP timeouts at defaults, login to your ATA (or Obitalk.com, depending on what you use), and navigate to Voice Services-->X_KeepAliveExpires. Change X_KeepAliveExpires to 35 seconds. If you're going to do this, then also navigate to Service Providers-->ITSP Profile used for FPL-->RTP-->KeepAliveInterval. Change KeepAliveInterval to 35 seconds as well.

Then navigate to Service Providers-->ITSP Profile used for FPL-->SIP-->RegisterRetryInterval. Change RegisterRetryInterval to 185 seconds.

Making those changes satisfies these conditions:

UDP Unreplied Timeout (in your router) < NAT Keep-alive Interval (in your ATA; for Obihai ATAs this is X_KeepAliveExpires; for Grandstream, the setting is SIP OPTIONS Keep Alive Interval) < UDP Assured Timeout (in your router) < SIP Registration Failure Retry Wait Time (or RegisterRetryInterval in Obihai ATAs)

“<“ means less than.


1) UDP Unreplied Timeout is 30 seconds by default in Asus routers < X_KeepAliveExpires is now 35 seconds < UDP Assured Timeout is 180 seconds by default in Asus routers < RegisterRetryInterval is now 185 seconds.


In the PDF guide, in order to conform to what's listed at https://support.freephoneline.ca/hc/en- ... redentials, you're told to make changes to UDP timeouts in your router (login to router, and navigate to "General-->Tools-->Other Settings") to change "UDP Timeout: Assured" to 115 seconds and "UDP Unreplied Timeout" to 15 seconds.

2) UDP Unreplied Timeout is 15 seconds < X_KeepAliveExpires is 20 seconds (based on Freephoneline's recommendations) < UDP Assured Timeout is 115 seconds in your router < RegisterRetryInterval is 120 seconds (based on Freephoneline's recommendations)

RegistryRetryInterval is the length of the time your ATA will take to attempt another registration with FPL's proxy server after registration fails. So, if you use 185 seconds for RegistryRetryInterval (to keep your router's UDP Assured Timeout at default), then the ATA will take an additional 65 seconds (185-120=65 seconds) before attempting to register again.

Both scenarios work, at least, for FPL.

Increasing X_KeepAliveExpires and KeepAliveInterval increases the risk of not keeping a UDP connection alive. So, generally, increasing those values isn't advisable, but 35 seconds does work without issue.

Increasing RegistryRetryInterval has the disadvantage of the ATA taking longer to retry registration after a failed registration attempt, but if you start decreasing RegistryRetryInterval below 120 seconds, you run the risk of getting a temporary IP ban due to too many registration attempts within a short interval. So, 120 seconds or higher is fine.


Lastly, if you are using the PDF guide and have setup server failover in the ATA, then
login to your router’s web UI.
Navigate to Advanced Settings–>Administration–>System (tab)–>Basic Config–>
Change “Enable WAN down browser redirect notice” to No.
Click “Apply

I did not have to disable Two-Way IPS in Merlin firmware version 386.7_2 for SIP URI calls. Possibly other firmware versions may require Two-Way IPS (Navigate to General–>AI Protection–>Network Protection) to be disabled.

Thank you again for all your help and if I have problems in the future I now know how to trouble shoot my way through most of them.


That's amazing! That's what I like to see.

If you do ever notice that you're getting packet loss (especially with that WinMTR test within the first two hops--or at the very last hop), contact Rogers and ask for a senior level tech (tier 2) to check for signal noise in your area and modem signal levels. Tier 2 has access to more diagnostic tools than the first rep you'll speak to. Check to ensure the coaxial cable is connected firmly to the back of the gateway that Rogers gave you.
Some modems are better able to handle noise and out-of-spec signal levels than others.

Thank you very much for reporting back.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk
Technical Support
 
Posts: 2763
Joined: 04/26/2010
SIP Device Name: Obihai 202/2182, Groundwire
Firmware Version: various
ISP Name: FTTH
Computer OS: Windows 64 bit
Router: Asuswrt-Merlin & others


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