phone cuts out @ 15 minutes

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cannot call

Postby M3231 » 08/29/2020

I get dial tone,
dial the number
all I get is BEEP! BEEP! BEEP!
(tried several numbers)
unplugged the VOIP modem, (and ethernet cord)
plugged it back in, let it reset, same thing.
any suggestions?
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Re: cannot call

Postby Liptonbrisk » 08/29/2020

Are you using Fongo Home Phone? If so, submit a ticket: https://support.fongo.com/hc/en-us/requests/new.

By the way, 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.

If you're using Freephoneline, specify

1) the brand and model modem you're using,
2) the brand and model router you're using,
and
3) the brand and model ATA you're using. If it's a Linksys/Cisco ATA, ensure that you've not accidentally enabled Caller ID block on your ATA. Dial *68 to remove caller ID blocking on all outbound calls.
Don't use "Anonymous" for display name in your ATA. Don't use or accidentally dial *67. In a linksys/cisco ATA, look for Voice-->Line (whichever one you're using for FPL)-->Supplementary Service Subscription-->Block CID Serv: change to no, and click "submit".
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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Re: cannot call

Postby M3231 » 08/30/2020

it started working yesterday after being unoperational for a few days so it can only conclude it is something on your end as we have had no internet interruptions during this time.

2) when we do make a call, it cuts off after 10 minutes without warning. This is not low battery, it just cuts off. Sometimes it will make it to 15 minutes.

3) I did submit a ticket but I got a response saying it would cost $50.00 to talk to a technician
a) I would not pay $50.00 to talk to a technician
b) the call would likely get cut off and Id still get charged.
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Re: cannot call

Postby Liptonbrisk » 08/30/2020

M3231 wrote:it started working yesterday after being unoperational for a few days so it can only conclude it is something on your end


None of the volunteer community moderators here work for nor represent Fongo or Freephoneline.
These are, mostly, user to user support forums. Support staff is not obliged to respond here (and neither is anyone else).

No; there was no service outage at the time you posted. Not losing internet connectivity does not mean the issue isn't on your end.
Most issues are on the user's end.



when we do make a call


My calls aren't dropping.

Given you haven't provided the information I requested, much less stated what service you're using, I can't help you.
I could guess, based on the response you received from your ticket, that you're using Freephoneline, but I'm not positive.

There is no free technical support for Freephoneline.
There is free technical support for Fongo Home Phone.


the call would likely get cut off and Id still get charged


https://support.freephoneline.ca/hc/en- ... al-Support
"If you purchase Pay Per Incident Technical Support, we will follow through with Telephone and/or Remote Desktop Support until your issue is 100% resolved.

Payment is required before we begin work on your issue. Refunds for this service are only applicable if the issue was determined to be the fault of 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.
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phone cuts out @ 15 minutes

Postby M3231 » 09/08/2020

everytime Im on a call, the fongo free phone line cuts out.

- I have one of the fastest internet plans available from Rogers about half a gig download speed.
- I had the modem replaced by Rogers
- reset the VOIP modem
- handsets are new panasonic and charged

sometimes Im on hold with tech support, cannot get through to anyone, then the phone cuts out and I have to start again.
this happens EVERY time after 10-15 minutes.
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Re: phone cuts out @ 15 minutes

Postby Liptonbrisk » 09/09/2020

I still have no clue whether you're using Fongo Home Phone or Freephoneline, which are separate and distinct services.
But Hitron CGN3ACSMR and CODA-4582 series gateway modem/router combos from Rogers (and possibly other ISPs) requires users to do the following:

Open the 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.
Contact your ISP if need be.

If you're using freephoneline, try "voip4.freephoneline.ca:6060" without the quotation marks for the proxy server.
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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Re: phone cuts out @ 15 minutes

Postby M3231 » 09/11/2020

Freephoneline howver to log into my account I must go though fongo, so I don't see how they are distinctively different.
2) the "free phone" is now working but cuts out after 10 minutes. is there some setting for this?
3) the "softphone" has never worked. Is there a link to re-download that? maybe something is corrupt?
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Re: phone cuts out @ 15 minutes

Postby Liptonbrisk » 09/11/2020

M3231 wrote:Freephoneline howver to log into my account I must go though fongo, so I don't see how they are distinctively different.


To log into any Freephoneline account, the user visits https://www.freephoneline.ca/login. If you're visiting fongo.com, I'm not sure what service you're referring to, but it's not Freephoneline.

Fibernetics is the parent company of Freephonline and Fongo. Fongo is a Fibernetics brand.
Fongo Home Phone, Fongo Mobile, and Freephoneline are all separate services.


the "free phone" is now working but cuts out after 10 minutes. is there some setting for this?


The most common causes are SIP ALG, lack of QoS, or an issue involving your ISP that is causing jitter or packet loss.


1) What brand and model router are you using?

If you don't have your own router and are just using a Rogers Hitron modem/router combo, then

a)Open a 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.
Contact your ISP if need be.

b) If you do have your own router, stick the Hitron modem/router combo in bridge mode: https://www.rogers.com/customer/support ... ridgemodem



2) What brand and model ATA or IP phone are you using?

a) use "voip4.freephoneline.ca:6060" without the quotation marks for the proxy server.

3) If you have your own router, refer to your manual, and enable QoS properly for your ATA or IP phone.


the "softphone" has never worked. Is there a link to re-download that? maybe something is corrupt?


The instructions here work: viewtopic.php?f=8&t=19063&p=74810#p74810.
The desktop app has always worked for me and continues to work using Windows 10 64 bit version 2004 (OS Build 19041.508).

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 the Freephoneline desktop app. 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.
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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Re: phone cuts out @ 15 minutes

Postby Liptonbrisk » 09/11/2020

For the Freephoneline desktop app . . .

Make sure that you're not muting anything (microphone/speakers), and that you tested to ensure your mic is working before fiddling around with the app: http://win10faq.com/fix-microphone-settings/

And make sure you test incoming calls for 1-way audio issues before paying anything to FPL (you'll need a mic and headphones/speakers to test). Test on a computer that's connected to your router (without DMZ or port forwarding enabled). Should you encounter 1-way audio issues, look for a feature called SIP ALG in your router (you may need to call your ISP if you're using a modem/router combo) and disable that feature.

Steps i,ii, and iv below are for help dealing with 1-way audio issues with Freephoneline desktop application.


from http://forums.redflagdeals.com/fongo-at ... #p27011164

You can try the Freephoneline desktop app for free: https://www.freephoneline.ca/downloadDesktopApplication
It requires 32-bit Java to run. If you have problems installing the desktop app, visit viewtopic.php?f=8&t=19063&p=74810.


A.Use winmtr https://sourceforge.net/projects/winmtr/

B. For Freephoneline.ca (based in Ontario), test to voip.freephoneline.ca (let winmtr ping about 100 times), voip2.freephoneline.ca, and voip4.freephoneline.ca. You can copy text to clipboard and paste your results (do not post your own IP public address though) and post them for others to examine if you want.

C. Look at the very last hop or line. Take a look at your average ping--and your maximum. You want those values to be relatively close.
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), you should probably avoid FPL.

I get between 11 (voip.freephoneline.ca and voip2.freephoneline.ca)-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.

Try the free FPL desktop app first: https://www.fongo.com/app/desktop/

Make sure that you're not muting anything (microphone/speakers), and that you tested to ensure your mic is working before fiddling around with the app: http://win10faq.com/fix-microphone-settings/

And make sure you test incoming calls for 1-way audio issues. Test on a computer that's connected to your router (without DMZ or port forwarding enabled). Should you encounter 1-way audio issues, look for a feature called SIP ALG in your router (you may need to call your ISP if you're using a modem/router combo) and disable that feature.


i. Typically it's better to have your own router and to stick whatever modem/router combo your ISP gives you into bridge mode.

ii. Disable SIP ALG in your own router. Many modem/router combos that are issued by ISPs have faulty SIP ALG/SPI functions enabled, with no way to disable them. These features can mangle SIP headers. If you don't know how to disable SIP ALG, contact your router's brand or contact your ISP.

To understand why SIP ALG is often a serious headache visit https://www.voip-info.org/routers-sip-alg/ (scroll down to "SIP ALG Problems")

iii. Properly enable QoS in your router for your computer that's running the Freephoneline desktop app (and ensure no other programs are running on your computer that are hogging bandwidth while using the Freephoneline desktop app). Refer to your router's manual or contact your ISP if you were issued a modem/router combo from them (typically those routers suck and have horrible or absent QoS functions).

I'm not a huge fan of this website, but it suffices for an explanation of QoS: http://www.voipmechanic.com/qos-for-voip.htm
Avoid anything it says about the G.729


iv. If you still get one-way audio issues with the Freephoneline desktop app, you may need to port forward, which is a security risk (and not advisable).

The FPL desktop app uses ports 5060-5061,6060-6061,13000-13001 if you're going to port forward for the desktop app (you need to port forward to the LAN IP of the computer you're using. For most home networks the IP will begin 192.168.xxx.x). Refer to your router's manual to learn how to port forward (if your router came from your ISP, contact your ISP).

I would start just by port forwarding 13000-13001 only, which is for RTP (audio packets). If that still doesn't work, you can try adding 6060 or 6061. The most dangerous ports to forward are 5060-5061 and really shouldn't be necessary if you're forwarding 6060 or 6061 anyway. I guess if all else fails, forward all of them: 5060, 5061,6060, 6061,13000, and 13001.

These are all UDP ports.

5060, 5061, 6060, and 6061 should be alternate SIP ports.

Only port forward if all else fails (and only do it temporarily, since it's a security risk).
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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Re: phone cuts out @ 15 minutes

Postby M3231 » 09/11/2020

thanks this is a lot to take in. I hope it's helpful to others as well.

My VOIP is CISCO SPA122 ATA with Router
modem is from Rogers Hitron
Model CODA 4582U

-- nobody shares the account
-- SIP ALG was under "Gateway Function" and it was disabled
-- canot bridge the modem. says it will cut off our wifi.
M3231
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Re: phone cuts out @ 15 minutes

Postby Liptonbrisk » 09/11/2020

M3231 wrote:
My VOIP is CISCO SPA122 ATA


By the way, the WAN to LAN throughput for the router in the SPA122 is limited to 20 Mbps, so it's not worth trying to use the SPA122 as a router for other devices on your local area network (LAN). It's too slow.

Please follow the steps listed, step by step:

1) Ensure that you've not accidentally enabled Caller ID block on your ATA. Dial *68 to remove caller ID blocking on all outbound calls.
Don't use "Anonymous" for display name in your ATA. Don't use or accidentally dial *67.

Login to your ATA. Navigate to Voice-->Line (whichever one you're using for FPL)-->Supplementary Service Subscription-->Block CID Serv: change to no, and click "submit".
For some reason, people have had calls drop or simply not work with that setting enabled.

Block CID Serv set to yes tries to make caller ID on all of your outgoing calls appear as unknown or anonymous to the recipient. And that setting no longer works properly with Freephoneline and causes problems with outgoing calls when it is enabled.


Should you still experience call drops after following these steps, the next time the problem occurs, log in at https://www.freephoneline.ca/callLogs. Select the current month. Look at the call you made, and check the "Disconnect reason" listed. Also, it might help to know if you're calling a regular landline, mobility number, or another VoIP number. In other words, does the problem happen with all calls or only with certain numbers?


2) Login to your ATA. Specify a high random SIP port in your ATA between 30000 and 60000.
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.
Do not use the same random SIP port for any other Line. Always choose a different random local SIP port for each Line you're using.

Using a high random SIP port may help to bypass SIP ALG, and it also helps to avoid SIP Scanners (or hackers).
Also, changing local SIP port will reset a potentially corrupted NAT association that developed between your router and ATA.

Click submit.

3) In the ATA, navigate to 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
d) NAT Keep Alive Interval--> 20 seconds

d) click "Save Settings" button

This helps to ensure the RTP audio stream is being sent to your WAN IP as opposed to your LAN IP.

4. Navigate to Voice-->Line (whichever you use for FPL)-->NAT settings
a) NAT Mapping Enable should be yes
b) NAT Keep Alive Enable should be yes
c) NAT Keep Alive Msg should be $NOTIFY

d) click "Save Settings" button if changes were made

5) In your ATA, navigate to Voice-->Line-->Proxy and Registration-->Register Expires needs to be 3600 seconds (it probably already is set to 3600)

6) Navigate to Voice-->SIP-->SIP Timer Values (sec)
Reg Retry Intvl should be 120 seconds

Click "Save Settings" button if changes were made

Many older guides for FPL don't include this setting.

7) 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. You don't have a separate router, but if you eventually get one, you know what to do.

a) You might also want to give "voip4.freephonline.ca:6060" (without the quotation marks) a shot in your ATA if problems persist. Remember 6060 is the proxy server port--and not the (local) sip port listed under SIP Settings in the ATA. In your ATA, you would navigate to Voice-->Line-->Proxy and Registration-->Proxy (use "voip4.freephonline.ca:6060" without the quotation marks).


modem is from Rogers Hitron
Model CODA 4582U


Okay, that is a modem/router combo or gateway. It's not just a modem; it also has a router built into it.
The modem portion of the device does not cause problems with FPL. But the router portion can.

SIP ALG is a router feature. It can cause problems. Ensure it's disabled:
Open a 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.
Contact your ISP if need be.



SIP ALG was under "Gateway Function" and it was disabled


Okay, great.


canot bridge the modem. says it will cut off our wifi.


I only meant for people to enable bridge mode if they're using their own router (other than the one given to them by their ISP). For example, Asus RT-AX88U is a consumer router.
Enabling bridge mode doesn't apply to you here.

And, unfortunately, the Hitron 4582 doesn't have a QoS feature (refer to #3 below).

nobody shares the account


My only point is that if you use Freephoneline's desktop app simultaneously while your ATA is on and while using the same FPL account, you can expect issues, in particular, with incoming calls.
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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Re: phone cuts out @ 15 minutes

Postby Liptonbrisk » 09/11/2020

(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 17 for UDP Unreplied Timeout and 117 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.
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Liptonbrisk
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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
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