Incoming calls blocked by Telus ISP senior plan?

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Incoming calls blocked by Telus ISP senior plan?

Postby carniver » 12/05/2022

I've moved onto Telus senior's internet plan for $20, ever since that my freephoneline can no longer receive calls. There are no SIP messages for the incoming call, but making outbound calls works fine.

My ATA is a Grandstream HT801. I've moved this unit to another home with Telus internet and it can receive calls just fine. I've also brought a working ObiHai OBI200, with another freephoneline account, to my home. It would not be able to receive calls at my home. Swapped my account onto the Obihai, no received calls. So the problem is with my Telus internet, but how can I prove it to Telus? They've been sending me flyers in the mail to sign up for their landline for $30 a month. Is this blackmailiing?
carniver
Just Passing Thru
 
Posts: 4
Joined: 08/11/2022
SIP Device Name: Grandstream HT801
Firmware Version: 1.0.41.2
ISP Name: Telus
Computer OS: Windows 10
Router: Telus router

Re: Incoming call blocked by Telus?

Postby Liptonbrisk » 12/05/2022

I'd be surprised if Telus, Bell, or Rogers (or their subsidiaries) started filtering/blocking UDP port 5060 on their network on purpose
But the hubs or gateways they use can create problems at default settings due to SIP ALG being enabled (and, in some cases, there's no way for the user to disable it), depending on the device's firmware that's used.
Moreover, you visited another household with Telus, and your ATAs work normally.


The comparison between households matters if the model of the Telus Hub or modem/router combo is exactly the same, with the same firmware version, and if the settings in them are the same as well.
SIP ALG is a router feature that can cause problems with incoming calls.
If you're using a modem/router combo, gateway, or hub issued by your ISP (and are NOT using your own additional separate router), contact your ISP to ask for assistance for disabling SIP ALG in the modem/router combo, gateway, or hub. Disable SIP ALG. Alternatively, try voip4.freephoneline.ca:6060, which is used to help circumvent SIP ALG.

I suspect the difference is SIP ALG in the router (or some router NAT feature) or Wi-FI hub being used. If you want to test this theory briefly, for Telus Wi-Fi Hubs, put LAN port 1 on the Hub in bridge mode. Then connect the ATA directly to LAN Port 1 on the hub. Only do this briefly while testing for incoming calls because the ATA will be unprotected while the hub is in bridge mode. Afterwards, revert the Telus Hub back to how it was previously.
Refer to point B from viewtopic.php?f=8&t=20199#p78976 to learn why SIP ALG can cause problems. Contact Telus if you need help enabling and disabling bridge mode.

Also, registration is required for incoming calls but not for outgoing calls. Check "SIP status" after logging in at https://www.freephoneline.ca/showSipSettings
Please note that if "SIP User Agent" does not reflect a device you're using, someone else is using your Freephoneline VoIP unlock key.

Lastly, you can't register the same FPL account on multiple devices simultaneously. Only the most recently registered device will work for incoming calls.
Only one registration (which includes a single FXS Port or SP on an ATA) is allowed per FPL account at any time.


--
The Obihai ATA you have, for a single line, is an easier device (and more powerful for use with FPL) to troubleshoot problems with than the Grandstream ATA.
For 1-way call issues, refer to "Are you 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)?" on pages 40 to 42 from the PDF guide, which can be found at the bottom of the first post from viewtopic.php?f=15&t=18805#p73839.

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. Login to ATA. 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.

Refer to page 10 of the guide if you want to use the Obitalk website. You need to enter the expert menu if you use the Obitalk.com website.

1. Disable SIP ALG in whatever router you’re using. Contact your ISP if need be.

2. In your Obihai ATA or at Obitalk.com, 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.
By using a high random port you help to thwart SIP scanners and may also circumvent a faulty SIP ALG feature in
your router. If you've already chose a random number in that range, choose a different number in that range.

3. 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.

4. Navigate to Service Providers-->ITSP Profile used for FPL-->SIP-->ProxyServer
a) use voip4.freephoneline.ca
b) change ProxyServerPort to 6060 for voip4.freephoneline.ca



submit/save settings

5. Navigate to Voice Services-->SP Service used for FPL
a) ensure X_KeepAliveEnable is Checked
b) X_KeepAliveExpires is 20
c) X_KeepAliveMsgType is notify

submit/save settings.


6. This is always proper device reboot order:

A.Turn off modem, router and ATA (or IP Phone or close SIP App).

B. Turn on modem. Wait for modem to be fully up and running.

C.Turn on router.
Wait for modem to be fully up and transmitting data before turning on router.

D. Turn on ATA (or IP Phone or open SIP app) only after the router is fully up and running.

The SIP device (ATA, IP Phone, or SIP app) should be the last device powered on in the device chain.


Reboot/power cycle ATA


7. Dial ***1. Enter the IP address you hear into a web browser. Login to the ATA.

a) Navigate to Status--System Status-->SP (Freephoneline) Service Status-->Status

The ATA needs to be registered for incoming calls to work.

8. Test incoming calls. Login at https://www.freephoneline.ca/callLogs and provide, especially, the "Disconnect reason".
Check and confirm that the "To" and "From" fields represent numbers you expect. Are you reaching FPL's voicemail system only? Do you hear anything?

9. If incoming calls are reaching FPL, then check if they're also reaching your ATA. Dial ***1.
Enter the IP address you hear into a web browser. Login to ATA. Navigate to Status-->Call History.
Do you see a 3 digit SIP error code on the right hand side?

--

For your Grandstream ATA, ensure that
"SIP REGISTER Contact Header Uses" is set to "WAN address".
For primary SIP server, use "voip4.freephoneline.ca:6060" without the quotation marks.
Use Random SIP Port should be Yes
Enable SIP OPTIONS/NOTIFY Keep Alive: NOTIFY
Keep Alive Interval should be 20 seconds

Check your other settings; double check against what's in this guide: viewtopic.php?f=15&t=20252#p79162. However, Subscribe for MWI should be set to No. Freephoneline sends unsolicited MWI.
Navigate to Status-->Port Status-->Registration to check registration status.
ATA needs to be registered for incoming calls to work.

--
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 FXS Port on your Grandstream ATA or SP on your Obihai ATA). 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).
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: Incoming call blocked by Telus?

Postby Liptonbrisk » 12/05/2022

If you're more focused on your ISP being to blame, then it would be useful to check for packet loss, but this isn't the problem if outbound calls work. What's below isn't useful for you at this time (but it might be at some point later).

----
The instructions below are for testing to FPL's proxy servers, but I would also test to 208.85.218.146 and 208.85.218.147, 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)
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.
User avatar
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: Incoming call blocked by Telus?

Postby carniver » 12/05/2022

Liptonbrisk wrote:3. 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)

Thanks, tried that, still no go

Liptonbrisk wrote:Please note that if "SIP User Agent" does not reflect a device you're using, someone else is using your Freephoneline VoIP unlock key

It's my ObiHai 200 indeed

Liptonbrisk wrote:4. Navigate to Service Providers-->ITSP Profile used for FPL-->SIP-->ProxyServer
a) use voip4.freephoneline.ca
b) change ProxyServerPort to 6060 for voip4.freephoneline.ca

That sounded promising, but still same thing, makes me start to question my sanity

Liptonbrisk wrote:5. Navigate to Voice Services-->SP Service used for FPL
a) ensure X_KeepAliveEnable is Checked
b) X_KeepAliveExpires is 20
c) X_KeepAliveMsgType is notify

Also same thing

Liptonbrisk wrote:8. Test incoming calls. Login at https://www.freephoneline.ca/callLogs and provide, especially, the "Disconnect reason".
Check and confirm that the "To" and "From" fields represent numbers you expect. Are you reaching FPL's voicemail system only? Do you hear anything?

Reached the voicemail after like 20 seconds, disconnect reason "Normal call clearing"
That's odd

Liptonbrisk wrote:9. If incoming calls are reaching FPL, then check if they're also reaching your ATA. Dial ***1.
Enter the IP address you hear into a web browser. Login to ATA. Navigate to Status-->Call History.
Do you see a 3 digit SIP error code on the right hand side?

No record of the incoming call, so no errors either

Liptonbrisk wrote:For your Grandstream ATA, ensure that
"SIP REGISTER Contact Header Uses" is set to "WAN address".
For primary SIP server, use "voip4.freephoneline.ca:6060" without the quotation marks.
Use Random SIP Port should be Yes
Enable SIP OPTIONS/NOTIFY Keep Alive: NOTIFY
Keep Alive Interval should be 20 seconds

Check your other settings; double check against what's in this guide: viewtopic.php?f=15&t=20252#p79162. However, Subscribe for MWI should be set to No. Freephoneline sends unsolicited MWI.
Navigate to Status-->Port Status-->Registration to check registration status.
ATA needs to be registered for incoming calls to work.

Grandstream doesn't seem to work with voip4.freephoneline.ca:6060 as the address, and I couldn't figure out how to change the port from 5060 to 6060.

.[/quote]

If you're using a modem/router combo, gateway, or hub issued by your ISP (and are NOT using your own additional separate router), contact your ISP to ask for assistance for disabling SIP ALG in the modem/router combo, gateway, or hub. Disable SIP ALG.

I'll try that now
carniver
Just Passing Thru
 
Posts: 4
Joined: 08/11/2022
SIP Device Name: Grandstream HT801
Firmware Version: 1.0.41.2
ISP Name: Telus
Computer OS: Windows 10
Router: Telus router

Re: Incoming call blocked by Telus?

Postby Liptonbrisk » 12/05/2022

carniver wrote:Thanks, tried that, still no go


I hope you're not reverting steps and setting changes on an individual basis. They are meant to be done in the order listed, cumulatively.

It's my ObiHai 200 indeed


Then at least the ATA is registering. Your Telus Hub doesn't seem to prevent all communication with FPL's proxy server.

disconnect reason "Normal call clearing"
That's odd


That's normal for reaching voicemail and disconnecting.

Ringing isn't heard; call doesn't reach ATA; call drops to voicemail.

I would have asked whether you have Do Not Disturb (sleep mode or the ringer turned off) enabled on the phone(s) attached to your ATA(s), except the Obihai ATA isn't even listing the incoming calls.

No record of the incoming call, so no errors either


The incoming call is reaching Fongo's network and your FPL account but not your ATA. I usually associate this problem with SIP ALG or a router NAT firewall/configuration issue. However, if you're using voip4.freephoneline.ca:6060 and setting X_UserAgentPort to 55555, for example, then the problem seems unlikely to be SIP ALG related. SIP ALG involves UDP port 5060, which isn't being used.

RTP audio stream from 208.85.218.146 and 208.85.218.147 is either not being sent to your public WAN IP address--or it's being blocked by your Telus Hub's firewall.
If both X_DiscoverPublicAddress is enabled and X_UsePublicAddressInVia are enabled, I think it's very unlikely that the RTP audio stream isn't being sent to your public WAN IP address (unless you're not letting me know you're using a separate router that's running a VPN).

That leaves me speculating about your Telus Hub's router firewall blocking incoming connections (or some firmware bug), but I would guess RTP IPs aren't being blocked if you hear incoming audio for outgoing calls.

I would want to quickly test with the Telus Wi-Fi Hub lan port in bridge mode. If it's in bridge mode, it's firewall and all router functions aren't being used. The hub would be acting as a modem only. Nothing can be blocked in bridge mode. Bridge mode (if you don't have your own router) is a huge security risk for your ATA since it won't be protected by the Telus Hub.


I'm also wondering whether Telus uses a different (problematic) hub for the senior plan than with faster plans.



Grandstream doesn't seem to work with voip4.freephoneline.ca:6060 as the address


It should, if the entire entry is listed for Primary SIP server.

Login to your ATA. Navigate to Status-->Port Status-->Registration to check registration status.

and I couldn't figure out how to change the port from 5060 to 6060


You shouldn't be changing local SIP port since "Use Random SIP Port" should be enabled. Local SIP (LAN) port is not the same as the proxy port, and it's understood when preceded by a colon for the primary sip server entry.
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.
User avatar
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: Incoming call blocked by Telus?

Postby Liptonbrisk » 12/05/2022

I'm wondering whether Telus uses a different (problematic) hub for the senior plan than with faster plans.

Are you able to find the brand and model of the device Telus gave you (possibly on the back or underside)? I don't use Telus, but maybe I can search for something useful.

Are you using a Telus hub? I know people that use a Telus Wi-Fi (white Arcadyan) hub with their own serparate router that I configured for them. LAN port 1 is in bridge mode. Everything works fine.

Or are you connected directly to the ONT? I've never tested that with Freephoneline.
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.
User avatar
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: Incoming call blocked by Telus?

Postby Liptonbrisk » 12/05/2022

I hope this isn't the problem: https://forum.telus.com/t5/Internet-Hom ... rue#M17876.

carlosfranco wrote:Telus rep said was because they blocked all the ports in the new Telus Modem last firmware.


That's a post from 2018, but even so, that can't be true???? All ports? Nothing would work.


Do you have an Actiontec T3200M?


https://www.manualslib.com/manual/11777 ... ml?page=63

Page 63 shows that you can disable SIP ALG.

Select "disable" for SIP, and click "Apply".

Menu is under "Advanced Settings".
https://www.manualslib.com/manual/11777 ... =40#manual

If a Telus Hub has a SIP ALG, they should be able to guide you to disable it.

Regardless if you followed earlier instructions properly for the OBi200 in this thread, it would be weird for the problem to be SIP ALG related (keep it disabled, regardless).

If you know other people using a different Telus hub don't have problems, then perhaps request that model hub.

carniver wrote:I've moved onto Telus senior's internet plan


Did they give you a different hub when you switched plans? Are you noticing intermittent internet outages?


carniver wrote: So the problem is with my Telus internet, but how can I prove it to Telus?



In response to your original question, I would probably point out that your SIP devices work with Telus at another location but not at yours. If you're on fibre, then you should be on the same Telus network. I would guess the only significant difference might be the hub.

Other than the Telus Hub being different, the other change would be reduced speeds and increased latency. However, you don't have a problem with outgoing calls, so I don't believe the reduced bandwidth matters.
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.
User avatar
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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