Freephoneline Connection Issue

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Freephoneline Connection Issue

Postby jsypun » 10/25/2020

Hi,

I had an issue with my FPL line, but it seems to have corrected itself out. Hopefully someone can let me know what caused the issue in the first place.

I was new to FPL back in mid-August. I have a Grandstream HT-802. When I first setup FPL back in August, I got a default phone number assigned to my account. I wanted to test out the service before I ported my number over from Bell. I tested it out for a week and the service was flawless. On my Grandstream HT-802, I noticed there were 3 icons on the ATA that lit up when everything was setup. All 3 icons were solid and none were blinking. I noticed that the icon for my cordless phone would blink when the phone was in use, but after I ended the call, the icon on the ATA went back to being solidly lit. Also, when I went to make an outbound call, the dial tone was a solid tone like you would normally hear when you make an outbound call on any landline.

Fast forward a week later and my number got ported from Bell at the end of August with no issues. It worked fine for a few days, but then I noticed around Sep 3rd or 4th an issue that persisted until today. When the phone was not in use, the icon on the ATA for the phone was blinking and not solid like before. Also, when I went to make an outbound call, the dial tone beeped for 10-15 seconds before sounding like a normal solid dial tone. I could still make outbound calls and I didn't necessarily need to wait for the dial tone to become solid before dialing out. I could dial out while the dial tone was beeping. I did notice the call quality was not as clear, but still acceptable. Also, sometimes when I received an inbound call, I couldn't hear the person on the the line. It sounded like I wasn't connected to them. They would hang up and call back and sometimes the issue would persist, but other times the call would come through. I did notice that when the call did come through, the line wasn't crystal clear as before. Again, the icon on the ATA for the phone was constantly blinking whether I was on a call or not. I did do a a soft and hard reboot on both my modem and ATA and initially for the first minute after the reboot the phone line icon on the ATA was solid and when I went to make an outbound call the dial tone was also solid like a normal dial tone with no beeping for that 10-15 seconds. After a minute though, the icon on the ATA went back to blinking and the dial tone on an outbound call was also beeping for the initial 10-15 seconds.

Today I woke up and thought about doing a factory reset on the ATA to see if that would resolve the issue. When I looked at the ATA, I noticed the phone line icon was solid and everything seemed back to normal. The FPL service is crystal clear and I am having no issues which is fantastic, but I am still puzzled as to what caused the issue. I have checked the status of FPL regularly and it doesn't seem they have been working on the service as everything on their reports seem fine.

One final note. I am using a Rogers combo modem/router which is directly connected to my ATA. I have had SIP ALG disabled on the Rogers modem/router since I first setup the service. I have been using voip4.freephoneline.ca/6060 as my primary SIP server since I am on Rogers and voip2.freephoneline.ca as my secondary SIP server when I noticed the issue. I had tried all 3 servers individually to see if it was a server issue, but the problem persisted on all 3 servers. So in the end, I still don't know what the issue was. Hoping someone can help in case this happens again in the future.

Thanks!
jsypun
Just Passing Thru
 
Posts: 5
Joined: 08/18/2020

Re: Freephoneline Connection Issue

Postby Liptonbrisk » 10/25/2020

jsypun wrote:the icon on the ATA for the phone was blinking and not solid like before. Also, when I went to make an outbound call, the dial tone beeped for 10-15 seconds before sounding like a normal solid dial tone.


Refer to pages 19 and 20 of your user manual for an explanation of LED patterns: http://www.grandstream.com/sites/defaul ... _guide.pdf.

Slow blinking is your visual voicemail indicator. The stutter dial tone indicates you have voicemail waiting.
When you receive voicemail, it may take up to 10 minutes for the indicator to appear. After you delete all voicemail, it may take up to 10 minutes for the indicator to disappear, including the stutter tone.

Sep 3rd or 4th


There was an issue involving incoming calls at that time (that I never experienced), but the problem was resolved shortly afterwards: viewtopic.php?f=8&t=19934&start=125#p78034.
That problem isn't related to the one you're describing concerning sound quality. If you're getting choppy audio or are intermittently (as opposed to constantly not hearing someone without rebooting your ATA) not hearing someone, it's likely a lack of QoS issue or an internet problem.

You may also want to check Preferred Vocoder settings in your ATA.
◦ Choice 1 should be PCMU unless you enjoy inferior audio quality

G.711u is PCMU or the equivalent of POTS (Plain Old Telephone Service)

G.729a is a lossy low bandwidth audio codec, which sounds vastly inferior, but uses less bandwidth than PCMU or G.711u. I would choose not to use the G.729a audio codec at all.




They would hang up and call back and sometimes the issue would persist, but other times the call would come through.



It could be a (lack of) QoS issue, an intermittent ISP issue at your home (signal issue causing random packet loss), a carrier issue, or something else.
You can help eliminate not having QoS as being the problem by using a decent router for SIP services and properly configuring QoS for your ATA.
Or you can ensure no other devices (other than your ATA) are connected via Wi-FI or ethernet cable to your router while making calls.


Grandstream Users (general settings)

i) For Primary sip server try "voip4.freephoneline.ca:6060" without the quotation marks.
6060 here has nothing to do with local sip port.

ii) Choose a random number between 30000 and 60000 for the local sip port.

Local SIP port is separate from the primary sip server port.

iii)Use random sip port should be set to yes.



Ensure

iv) Random RTP Port: Yes

v) SIP REGISTER Contact Header Uses is set to WAN address (if the setting is available)

vi) Register Expiration is 60 minutes

vii) SIP Registration Failure Retry Wait Time: 120 seconds

viii) Enable SIP Options Keep Alive: Yes

ix) SIP OPTIONS Keep Alive Interval: 20

x) Proper device reboot order is always modem (wait for it to be fully up before turning on your)-->router (ensure Wi-Fi SSIDs are populated first on your devices)-->ATA (wait for router to be fully up and running before turning on ATA). That's always proper device reboot order.

Lastly, ensure, after logging in at https://www.freephoneline.ca/showSipSettings that

i) SIP Status shows "connected", and
ii) SIP User Agent reflects a device that own and recognize. If you don't recognize the SIP User Agent, chances are you've been hacked.

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, and, consequently, incoming calls will not work on it. 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 or with another device. Registration is required for incoming calls. It is not required for outgoing calls. A more significant concern, though, is that multiple registration attempts can lead to temporary IP bans. The more devices being used can make the temporary ban happen more quickly. Note that each time you reboot or restart your ATA or SIP app, it's attempting to register with Freephoneline again. Multiple registration attempts within a short period can result in temporary IP ban. Each time you reboot your ATA it's attempting to register with FPL's proxy server.



I am using a Rogers combo modem/router


While some people rarely experience issues when not using a proper router for SIP services, not using a proper router with the characteristics described in the following post is really not an ideal situation.

Take a look at point #3 below and also #4.

If the issue just suddenly disappeared, it might have been a temporary ISP issue that resolved in your area. While I have no way of knowing, I think it's unlikely FPL suddenly switched interconnect carriers or anything else.
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: 1582
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:5921EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: Freephoneline Connection Issue

Postby Liptonbrisk » 10/25/2020

(Generic info)

Typically, for VoIP SIP services, especially for freephoneline, you want

1) a router that does not have a full cone NAT,

Visit https://www.think-like-a-computer.com/2 ... es-of-nat/.
Mango from the Obitalk.com forums writes,
“Use a restricted cone NAT router, and do not use port forwarding or DMZ. Restricted cone NAT will only permit
inbound traffic from the service provider you're registered to. If you have a full cone NAT router, it will allow traffic
from any source. This is probably not what you intend.
If you have a Windows computer, you can test your router using the utility here:
http://www.dslreports.com/forum/remark,22292023. To run it, use stun stun.ekiga.net from a command prompt.”
Essentially, you download the stun-test.zip file; extract the stun.exe file from within the zip file to an easily
accessible location; use an elevated command prompt (visit
http://www.thewindowsclub.com/how-to-ru ... inistrator); change directory (cd) to the
directory or location where you extracted stun.exe (visit
http://www.digitalcitizen.life/command- ... c-commands); and type “stun stun.ekiga.net” without
the quotation marks followed by the enter/return button on your keyboard.
Asus routers, at the time of this writing, produce port restricted cone NAT routers, for example and are fine,
provided you’re using one with Asuswrt-Merlin, third party firmware installed.

2) a router that lets you disable SIP ALG if it's buggy,

To understand why SIP ALG often causes horrible problems, please visit
https://www.voip-info.org/routers-sip-alg/ (scroll down to the section on SIP ALG problems).

If you're dealing with a modem/router combo issued by an ISP or a router with SIP ALG forced on, you may have
to use voip4.freephoneline.ca:6060 for the Proxy Server. The purpose of voip4.freephoneline.ca:6060 is to circumvent
faulty SIP ALG features in routers.

3) a router that allows you to set QoS or assign highest priority to your ATA or IP Phone over all other devices on your LAN (local area network),

For a very general description of what QoS can do for you, visit https://www.voipmechanic.com/qos-for-voip.htm.
The basic idea is if you're torrenting or have a bunch of other computers, smartphones, tablets, etc. downloading and uploading (hogging all your available bandwidth), you don't want
your ATA not to have access to enough bandwidth to make or receive calls properly. So QoS or a Bandwidth Monitor feature (which is just another form of QoS) is a really good idea for VoIP users.

I often get an occasional relative complaining to me, "Hey my calls sound choppy." And then when I go visit, some kids are playing MMOs on a computer, while another person is downloading a huge file,
and another person is backing up files to a cloud service all at the same time someone else is trying to talk on the phone. All those devices, without QoS enabled, are fighting over available bandwidth along with the ATA.

and 4) A router that lets you adjust both Unreplied and Assured UDP timeouts.

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 (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.

When a modem leases a new IP address, a problem can arise where prior associations using the old IP address are maintained in the router. When the ATA attempts to communicate using the old IP address, the response is unreplied, and then if the UDP Unreplied timeout is greater than the Keep Alive Interval (and UDP Unreplied timeout is often set to 30 by default in consumer routers) a problem arises where the corrupted connection persists. If UDP Unreplied timeout is, for example, 17, and the NAT Keep Alive Interval is 20, then the corrupted connection will timeout or close. A new connection will be created, and everything will work fine.

Another 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, X_Keepalives expires is supposed to be 20 with FPL.

(the above settings are making reference to those in Obihai ATAs)

Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers may be difficult, if not impossible. Asuswrt-Merlin (I would avoid any model below/less powerful than an RT-AC68U), third party firmware for Asus routers, does offer easy access to these two settings, which are found under General–>Tools-->Other settings. 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. Note that I will not be held accountable any damage resulting from failed firmware updates. Apparently, Mikrotik routers also allow users to change both Assured and Unreplied UDP timeout settings as well: https://forums.redflagdeals.com/recomme ... #p28056619.

Router firmware that allows users to adjust Assured and Unreplied UDP timeouts include

Asuswrt-Merlin
Ubiquiti
Mikrotik
pfSense
Tomato
DD-WRT


The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 15 for UDP Unreplied Timeout and 115 for UDP Assured Timeout.



ISPs do not issue customers routers that can do all four things I just listed. Typically it's far better to have your own router with strong QoS functions and a restricted cone NAT firewall,
disable whatever SIP ALG feature is enabled in the router, and stick whatever modem/router combo your ISP gives you into bridge mode. For Bell Hubs, visit http://forums.redflagdeals.com/please-s ... r-1993629/. For Rogers, visit https://www.rogers.com/customer/support ... ridgemodem.
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: 1582
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:5921EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: Freephoneline Connection Issue

Postby jsypun » 10/25/2020

Thanks Liptonbrisk.

Liptonbrisk wrote:
jsypun wrote:the icon on the ATA for the phone was blinking and not solid like before. Also, when I went to make an outbound call, the dial tone beeped for 10-15 seconds before sounding like a normal solid dial tone.


Refer to page 19 of your user manual for an explanation of LED patterns: http://www.grandstream.com/sites/defaul ... _guide.pdf.

Slow blinking is your visual voicemail indicator. The stutter dial tone indicates you have voicemail waiting.
When you receive voicemail, it may take up to 10 minutes for the indicator to appear. After you delete all voicemail, it may take up to 10 minutes for the indicator to disappear, including the stutter tone.



Out of curiosity, how would I get a voicemail if my cordless phone answering machine is set to less rings then my FPL voicemail. I purposely set it up that way. I have received a number of messages on my cordless phone voicemail which I have checked and deleted while I was having this issue. In fact I have some messages waiting on it right now which I have not checked yet. Also, does the FPL voicemail delete on its own after a certain number of days? I have never checked my FPL voicemail and the LED stopped blinking on it’s own on my ATA.
jsypun
Just Passing Thru
 
Posts: 5
Joined: 08/18/2020

Re: Freephoneline Connection Issue

Postby Liptonbrisk » 10/25/2020

jsypun wrote:
Out of curiosity, how would I get a voicemail if my cordless phone answering machine is set to less rings then my FPL voicemail.


When your ATA is not registered, or if an incoming call can not reach you for some reason (power outage if you don't have a UPS backup), the incoming call will be redirected automatically to Freephoneline's voicemail system. If you let an incoming call go unanswered while you're on an existing call (call waiting), the incoming call will also go to FPL's voicemail. In other words, it is impossible for anyone not to use Freephoneline's voicemail system at some point.

Also, does the FPL voicemail delete on its own after a certain number of days?


Yes. Please visit https://support.freephoneline.ca/hc/en- ... ges-expiry.

Voicemail settings can be found after logging in at https://www.freephoneline.ca/voicemailSettings.

By the way, registration is a requirement for incoming calls to work. While it's a good idea to check registration status in your ATA when you encounter problems, it's also a good idea to check SIP status after logging in at https://www.freephoneline.ca/showSipSettings. Note that because the registration interval for Freephoneline is 1 hour, SIP status at the website may not reflect your ATA's real registration status since it won't be updated until your ATA attempts registration with FPL's proxy server again (every hour).

I suggest not deleting voicemail from https://www.freephoneline.ca/mailbox. You may encounter this bug: viewtopic.php?f=8&t=17194&p=72630#p72630.
It's possible the problem doesn't occur anymore, but it's so annoying when it does that I don't want to test again:
viewtopic.php?f=8&t=17194&p=67866#p67850.
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: 1582
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:5921EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: Freephoneline Connection Issue

Postby jsypun » 10/27/2020

Thanks Liptonbrisk. You are always extremely helpful with your detailed explanations. Really appreciate it!
jsypun
Just Passing Thru
 
Posts: 5
Joined: 08/18/2020

Re: Freephoneline Connection Issue

Postby Liptonbrisk » 10/27/2020

You’re welcome!
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: 1582
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:5921EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin


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