Unable to call just one particular number

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Unable to call just one particular number

Postby Venturess » 06/03/2020

The strangest thing is happening. I can call any number on earth except one who also uses a Freephoneline number. When I call, I can't hear the phone ringing and neither can the other person, however, it eventually does kick into her voicemail. She can call me without a problem though. Calling the same number several times has never been a problem until about 5 days ago. Is there any explanation or do I need to do something on my end? All else has been phenomenal and I have been using Freephoneline for more than 8 years without a single issue! Has this happened to anyone else?

Thank you!
Venturess
Just Passing Thru
 
Posts: 4
Joined: 03/13/2014
ISP Name: Cable
Computer OS: Windows 7
Router: Linksys

Re: Unable to call just one particular number

Postby Liptonbrisk » 06/03/2020

If you can both receive and place calls with all other destinations fine, the problem is likely on your friend's end.
See here for an example: viewtopic.php?f=8&t=19773.

A) 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 incoming calls will not work on it. This is especially important to consider if someone else is using your friend's SIP credentials (username and password) that are found after logging in at https://www.freephoneline.ca/showSipSettings (or if she is trying to register her FPL account with a smartphone SIP app or with another device). Registration is required for incoming calls. It is not required for outgoing calls. Your friend needs to log in at https://www.freephoneline.ca/showSipSettings to check SIP Status and SIP User Agent. SIP Status needs to show connected. SIP User Agent needs to reflect the device she's using (her ATA) and not something else.


B) Ensure SIP ALG (pay close attention to #2 below) is disabled in the router or modem/router combo your friend is using. Alternatively, she can use voip4.freephoneline.ca:6060 for the proxy server.





If your friend wants help, she needs to provide
i) the brand and model of her modem,
ii) the brand and model of her router,
and
iii) the brand and model of her ATA or IP Phone.



---

(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.
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: 1311
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: Unable to call just one particular number

Postby Liptonbrisk » 06/03/2020

*continued*



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) < 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.

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.
User avatar
Liptonbrisk
Technical Support
 
Posts: 1311
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: Unable to call just one particular number

Postby Venturess » 06/04/2020

Thank you very much indeed for your instructions. I shall forward them and will pm or post when the issue is resolved. My sister has changed her Internet provider so it makes sense.
Venturess
Just Passing Thru
 
Posts: 4
Joined: 03/13/2014
ISP Name: Cable
Computer OS: Windows 7
Router: Linksys


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