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That password is not valid - Message

PostPosted: 02/06/2019
by pp0ff1
Hello,

My freephone has been working fine for years but now I am unable to make or receive phone calls. I have not made any changes and after resetting/rebooting all devices I get a message when trying to dial out - "That password is not valid". Also for incoming calls it goes straight to VM. I have been using voip4.freephoneline.ca:6060 and it says registered status on the ObiTalk page.

On my showSipSettings page in freephoneline.ca, it says disconnected for Sip Status.

I tried to contact support but they asked me to update the firmware or use pay per use support. I don't think this is an issue on my end. Can anyone provide any recommendation or guidance with this.

Thanks

Re: That password is not valid - Message

PostPosted: 02/06/2019
by Liptonbrisk
You may have been hacked. Don’t port forward, don't use DMZ (especially), and don't use UPnP unless you have no choice. Don't use default passwords (change them) for any device connected to the internet. Login at https://www.freephoneline.ca/showSipSettings. Copy and paste your SIP username and password into your Authusername and password settings. Double check to ensure no spaces are pasted before or after the password. Outbound calls won’t work if your SIP credentials are wrong. Incoming calls will go straight to voicemail if your ATA is unregistered.

Also if you've recently ported a number into Freephoneline, your SIP username would have changed.

Afterwards, check your settings against those in the PDF guide: viewtopic.php?f=15&t=18805#p73839. Using the default FPL profile at Obitalk.com can lead to problems. Especially refer to page 42 (about registration limitations) and page 47 (about lack of registration).

Re: That password is not valid - Message

PostPosted: 02/06/2019
by pp0ff1
Coping and pasting my password from showSipSettings page back into OBiTalk FPL setting fixed the issue. Thank you so much for your help. I will turn on DMZ as per your feedback. Thank again.

Re: That password is not valid - Message

PostPosted: 02/06/2019
by Liptonbrisk
pp0ff1 wrote:Coping and pasting my password from showSipSettings page back into OBiTalk FPL setting fixed the issue. Thank you so much for your help.


I'm glad the service is working for you. However, I suggest you go through the PDF guide fully to configure your ATA properly: viewtopic.php?f=15&t=18805#p73839. Using Obitalk.com's default Freephoneline profile can lead to problems.


I will turn on DMZ as per your feedback.


No. That's exactly what I don't want you to do. Enabling DMZ is the worst thing possible for security.

Don't use DMZ.
Don't use port forwarding, unless nothing else works.
Don't use UPnP, unless nothing else works.
Those are all security risks.
Change default passwords for devices connected to the internet.

Re: That password is not valid - Message

PostPosted: 02/06/2019
by Liptonbrisk
With respect to security, refer to #1 below.



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.