Changed to Rogers gateway

Have a question or problem with your Fongo Home Phone service? This forum is the place to get help from both staff and fellow community members.
Visit The Fongo Home Phone Support Knowledge-base for more.

Changed to Rogers gateway

Postby milo » 12/10/2020

Hi:
I have Grand Stream Handy tone -286 adapter for maybe last 10 years. System worked great. With freephone line and Fongo.
Today I switched to Rogers Ignite system with new router and my phone doesn't work. No dial tone.
When I am calling X number from my Fongo phone I am getting busy signal.
When I am calling Fongo phone from my cell I am getting busy signal.

Rogers is telling me that problem is on phone side. I do not understand. Worked with 2 different Routers.
I have no option but ask You for help.
Can You help please.

Richard
Last edited by Liptonbrisk on 12/11/2020, edited 2 times in total.
Reason: deleted email and phone number
milo
Just Passing Thru
 
Posts: 15
Joined: 10/08/2011

Re: Changed to Rogers gateway

Postby Liptonbrisk » 12/11/2020

The thread you posted in was unrelated to your issue, so I gave your post a new thread.
Also, posting your email address and phone number in a public forum is a great way to get spammed (or worse), so I deleted them for security reasons.
Fongo support staff is not obliged to read the forums nor respond here. If you want to reach them, submit a ticket: https://support.fongo.com/hc/en-us/requests/new.
In your ticket, request to be placed on an alternate Fongo proxy server due to SIP ALG.

Your issue is likely due to point #2 in the post below. Disable SIP ALG in the modem/router combo that Rogers gave you.

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

If you have an Arris broadband gateway,
open your web browser, and login at 192.168. 0.1
Navigate to Advanced-->Options.
Uncheck the SIP box.
Click "Apply".

When new firmware updates are pushed to Hitron devices from Rogers, settings may be reset. Check to ensure SIP ALG remains disabled, periodically, or especially when you have problems.
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: Changed to Rogers gateway

Postby Liptonbrisk » 12/11/2020

(Generic info)

Typically, for VoIP SIP services, especially for Freephoneline/Fongo, you want

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

Visit https://dh2i.com/kbs/kbs-2961448-unders ... -punching/.
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: 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


Return to Community Support

Who is online

Users browsing this forum: Majestic-12 [Bot] and 5 guests