[SOLVED] Outgoing call to one specific phone number fails

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[SOLVED] Outgoing call to one specific phone number fails

Postby goto78 » 05/11/2020

Hi,
I'm using long-distance calling cards for many years. But it's been about four days my FPL cannot reach its phone service anymore. After I dialed 514-447-0555 or 613-366-0555 or 416-477-5555, I waited for a few seconds then I heared the hang up tone. I haven't modified any configuration in my VoIP device, nor in my modem/router before that.
But it's working when I used my cell phone with another company to dial the same phone numbers, I heared the calling card phone system asking to enter a code.
Is anyone experiencing the same problem with FPL? Or is it only me?
goto78
Just Passing Thru
 
Posts: 2
Joined: 05/11/2020
SIP Device Name: Linksys PAP2T
Firmware Version: 3.1.15(LS)
ISP Name: CIKTelecom
Computer OS: Windows 10 & Ubuntu 18.04
Router: KW5262C15A

Re: Outgoing call to one specific phone number fails

Postby Liptonbrisk » 05/11/2020

goto78 wrote: Or is it only me?


I can't reproduce your issue.

"Please enter your authorization code now."

If you want help, provide

1) the brand and model of your modem, and
2) the brand and model of your router.

Regardless, a good starting point would be following steps 1 to 12, step by step, towards the bottom of this post: https://forums.redflagdeals.com/freepho ... #p26808549.
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: 1247
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: Outgoing call to one specific phone number fails

Postby goto78 » 05/11/2020

Thanks @Liptonbrisk. But I can't find any post saying something about steps 1 to 12 from your RFD link. I believe it's actually this link : viewtopic.php?t=19773&p=77219#p77220

It's fixed. I followed that link and changed in my ATA:
-SIP Port to a random number between 30000 and 60000 (e.g. 45782) (step 4)
-VIA parameters (step 6)
and I deleted STUN server field (step 8).

Then, for fun, I tried to revert back to my original settings but I can't reproduce the issue anymore. Strange...
goto78
Just Passing Thru
 
Posts: 2
Joined: 05/11/2020
SIP Device Name: Linksys PAP2T
Firmware Version: 3.1.15(LS)
ISP Name: CIKTelecom
Computer OS: Windows 10 & Ubuntu 18.04
Router: KW5262C15A

Re: Outgoing call to one specific phone number fails

Postby Liptonbrisk » 05/11/2020

goto78 wrote:Thanks @Liptonbrisk. But I can't find any post saying something about steps 1 to 12 from your RFD link.


They're listed under the section called "Full Steps" from https://forums.redflagdeals.com/freepho ... #p26808549.
The link I posted is correct. Anyway, I wasn't aware that you were using STUN, so it's fortunate you came across the other link.


Then, for fun, I tried to revert back to my original settings but I can't reproduce the issue anymore. Strange...


NAT corruption can develop in routers without users doing anything. Sometimes rebooting the router and ATA (in that order) can address that problem, as well as changing local SIP Port (for registration issues).
Afterwards, the problem won't reoccur until NAT corruption develops again. A high random local SIP Port also helps to thwart SIP Scanners (hacker scripts) that try to break into your devices and services through UDP ports 5060 and 5061 (common local SIP ports). Concerning general UDP timeouts/NAT issues, refer to the post below.

The VIA settings help for one-way audio issues when using FPL.

STUN should be avoided if possible since using STUN introduces an additional point of failure. In the event the STUN server drops, so does your SIP (FPL) service.
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: 1247
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: [SOLVED] Outgoing call to one specific phone number fail

Postby Liptonbrisk » 05/11/2020

(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) < 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: 1247
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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