user is not available,, No caller ID

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user is not available,, No caller ID

Postby edaou » 01/08/2019

Hello
I am a Freephone line user since 2013 (SPA122 Device) and everything was working fine without any problem
Two days ago I start facing a problem,
1) When I am trying to call my FPL number always I am getting user you are calling is not available,
2) When I am calling from my FPL number you can establish a normal call (two way communication), but on the other end you will see No Caller ID instead of your FPL number.
3) when checking my account on the FPL webpage under Status: You can see the device is registered

I Reset the device three time, still the same problem
I have a REAL-Public IP on My device, No-Nat or port translation

can someone help me on this
thank you in advance
Regards,
edaou
Just Passing Thru
 
Posts: 3
Joined: 07/13/2013
SIP Device Name: pap2t
Firmware Version: 5.1.6
ISP Name: gemstelecom
Computer OS: windows
Router: lynksys

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Re: user is not available,, No caller ID

Postby Liptonbrisk » 01/08/2019

edaou wrote:1) When I am trying to call my FPL number always I am getting user you are calling is not available,


1. What brand and model router are you using?

2. What brand and model modem are you using? I just want to know if you're on cable or dsl.

3. Are you given the option to leave a message? That is, are you reaching Freephoneline's voicemail system immediately on incoming calls?

4. Have you checked to ensure you have not accidentally enabled Do Not Disturb (DND) in your ATA?

Log into your SPA122
A. Under Voice-->User 1 and User 2 (both of these)-->navigate to Supplementary Service Settings

Ensure

a) DND setting is set to NO
b) Block ANC Setting is set to NO
c) DND Activated is set to NO
d) Cfwd All Serv: no
e) Cfwd Busy Serv: no
f) Cfwd No Ans Serv: no

B. Under Voice-->User 1 and User 2 (both of these)

Check Call Forward Settings


a) Cfwd All Dest:

That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.

b) Cfwd Busy Dest

That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.


c) Cfwd No Ans Dest

That field should be blank. If there's a phone number there you don't recognize, you were hacked.
Make that field blank.

5. When you check your call logs, are you seeing calls that you don't recognize? Login at https://www.freephoneline.ca/doGetCallLogs.
If so, someone else is using your service without your permission.

6. Are you using Follow Me? Is it enabled? If so, check your Follow Me number list to ensure there's not a number that you don't recognize:
https://www.freephoneline.ca/addFollowMeNumber.

a. Actually, you may want to completely disable Follow Me for testing purposes: login at https://www.freephoneline.ca/followMeSettings. Disable Follow Me.

7. You will hear the "user you are trying to reach is not available" message under these conditions:

a) FPL's voicemail is already engaged with a caller and another incoming call is routed to FPL's voicemail while it's already in use, or

b) a call is already in place, Follow Me is enabled; and someone else not dialing from an FPL, Fongo Mobile, or Fongo Home Phone number calls you.

c) If the second call is from a FPL, Fongo Mobile, or Fongo Home Phone number, then the call will be forwarded to the Follow Me phone numbers, and after not being answered after ringing the Follow Me numbers, the second call eventually gets dropped to the "The user you are trying to reach is not available" message.

FPL allows any combination of two simultaneous channels at any given time (1 outgoing and 1 incoming; 2 incoming; or 2 outgoing).
i. FPL to FPL, FPL to Fongo Mobile, FPL to Fongo Home Phone are all SIP URI calls.
ii. FPL doesn't normally allow inbound SIP URI calls from other VoIP networks, but it does within its related sister (Fongo) companies.
iii. SIP URI calls don't count as a channel being used.

When I have Follow Me enabled and am engaged in a call, I get the "user you are trying to reach is not available" message when there's an incoming call from anything other than an FPL, Fongo Mobile, or Fongo Home Phone number. So unless the second call is coming from a FPL, Fongo Mobile, or Fongo Home Phone number, then Call Waiting is not going to work if Follow Me is enabled. Basically, Follow Me = No Call Waiting unless the incoming call is from a Fibernetics VoIP call.

8. Check to ensure you're not using FPL's desktop app or using your Freephoneline SIP credentials on any other device while testing.

9. In your SPA122, 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.

10. In your SPA122, Navigate to the 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 Intvl: 20



11. Navigate to Voice-->SIP-->SIP Timer Values (sec)-->Reg Retry Intvl should be 120 seconds (this isn't related to your issue, but many people are missing this setting)



When I am calling from my FPL number you can establish a normal call (two way communication), but on the other end you will see No Caller ID instead of your FPL number.


a)Under Voice-->User 1 and User 2 (both of these)-->navigate to Supplementary Service Settings
Ensure Block CID Setting is set to no.

b) Navigate to Voice-->Line (FPL)--> Subscriber Information--> Display Name should be your first name and last name


If you want to test, use the free Fongo Mobile smartphone app and make calls to it: https://www.fongo.com/ using Freephoneline. Make sure you delete your FPL phone number in your smartphone's address book before testing. Outgoing CID info from FPL will appear in the Fongo Mobile app if your ATA is configured and working properly.


I Reset the device three time


Proper device reboot order is always modem-->router (wait for Wi-Fi SSIDs to populated first)-->ATA (in that order)


I have a REAL-Public IP on My device, No-Nat or port translation


I'm not sure what you mean by that. Where are you finding that information?

In your ATA, navigate to Voice-->Line -->NAT Settings

NAT Mapping Enable should be yes
NAT Keep Alive Enable should be yes

Under Voice-->SIP-->NAT Support Parameters
NAT Keep Alive Intvl should be 20
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: 848
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: user is not available,, No caller ID

Postby edaou » 01/08/2019

Thank you Very much for your help
Exactly as you said I was hacked , and under user-1 i saw that
1) DND setting was activated YES,
2) Block CID setting : yes also on the top of the page I saw the
3) Cfwd All Dest: + Cfwd No Ans Dest: + Cfwd Busy Dest: (There was an International number here in those Fields) .

user-2 has the default setting I Set the settings on user-1 like the User-2 and everything went to normal
I reboot the device and all is OK right now

Ps:
2) I disabled the Remote Access on thr Device
3) as for the Public IP
Let me share this with you
My VDSL modem is configured as bridge and the PPOE is set on my SPA122 (for that I am receiving the Public IP from the Internet-Provider directly on My-Cisco SPA Unit)

I have another Question : did freephoneline recommend StunServer if am behind a Firewall (this is not in my case but just asking)


Thank again for your help
Regards,
edaou
Just Passing Thru
 
Posts: 3
Joined: 07/13/2013
SIP Device Name: pap2t
Firmware Version: 5.1.6
ISP Name: gemstelecom
Computer OS: windows
Router: lynksys

Re: user is not available,, No caller ID

Postby Liptonbrisk » 01/08/2019

edaou wrote:Thank you Very much for your help


You're very welcome!

I have another Question : did freephoneline recommend StunServer if am behind a Firewall (this is not in my case but just asking)


No. STUN is an additional point of failure. If you really do need STUN, in the event the stun server drops, so does your service.

By the way, your device configuration guide can be found at viewtopic.php?f=15&t=16340#p64442.

My VDSL modem is configured as bridge and the PPOE is set on my SPA122


If that's actually a modem/router combo (or gateway) as opposed to just being a modem, by using PPPoE in the ATA, you're bypassing the router (including the NAT firewall) features in the modem/router combo (or gateway). In that case, you should make sure in the ATA, under Network Setup-->Basic Setup-->Network Service, that the Networking Service is set to NAT.

It's unfortunate that you were hacked, but I'm glad you discovered that you were instead of letting the problem persist.

A good firewall is useful as is ensuring that you're not using default passwords on any device that's connected to the internet. Port forwarding and, especially, using DMZ are security risks.
Point #1 below may be of interest to you.
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
Technical Support
 
Posts: 848
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: user is not available,, No caller ID

Postby Liptonbrisk » 01/08/2019

(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
http://www.voip-info.org/wiki/view/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 ... 2115672/2/

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: 848
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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