[Resolved] SIP Status: disconnected

[iceonu]

On the freephoneline.ca page under sip settings. I show:SIP Status: disconnected. Yet, my VoIP is online and working. Is this referring to the desktop app, which is no longer available?

[Liptonbrisk]

Is this referring to the desktop app

No. It refers to anything being used by your SIP username and password. Are you logged into the correct FPL web account?

My SIP status shows "connected"

And another account where I don't have a device connected indicates "disconnected".

which is no longer available?

It's still available: https://www.freephoneline.ca/downloadDesktopApplication.

[iceonu]

Are you logged into the correct FPL web account?/
Yes, I only have one FPL account. I haven't made any changes to it.

It's still available: https://www.freephoneline.ca/downloadDesktopApplication./
sorry, I should have said not available for Mac OS. But now I see that's irrelevant to my first question.

[Liptonbrisk]

Yes, I only have one FPL account. I haven't made any changes to it.

Registration (connected status) is a requirement for incoming calls but not for outgoing calls (unless "Outgoing Call without Registration" is set to "No" in the Grandstream device). Do incoming calls work?
If incoming calls work that always means you're registered.

Are the SIP username and password the same ones you use in your ATA?

Does reloading the web page change sip status? Can you try another browser?

If incoming calls work and if your browser is reporting current information accurately, then report the issue in a ticket: https://support.fongo.com/hc/en-us/requests/new.
There's nothing moderators and users can do about it. The SIPstatus should be read from their own server. I'm unable to reproduce this issue.

[iceonu]

Registration (connected status) is a requirement for incoming calls but not for outgoing calls (unless "Outgoing Call without Registration" is set to "No" in the Grandstream device). Do incoming calls work?
If incoming calls work that always means you're registered.

Incoming calls not working. Rebooted device, now showing not registered on device. No inbound/outbound calls.

Are the SIP username and password the same ones you use in your ATA?

Yes, SIP username displayed on FPL settings page are the same. Copy & Paste.

Does reloading the web page change sip status? Can you try another browser?

no change.

[Liptonbrisk]

Incoming calls not working.

Okay, then you're not registered, and FPL's website portal was correct.

Rebooted device, now showing not registered on device.

If your SIP device/app is configured properly, registration status is updated every hour. The registration interval should be 3600 seconds.
Registration status will not update until another registration attempt is made. This can be forced by rebooting SIP device because it attempts to register again.

In between each 3600 second interval, registration status may not be accurate. It will be accurate after rebooting the SIP device or after the SIP device attempts to register again.

viewtopic.php?f=8&t=20282&p=79316#p79307

Yet, my VoIP is online and working.

I'm uncertain what you mean by "working" if inbound calls aren't.

[Liptonbrisk]

Please go through the first post at viewtopic.php?f=8&t=20199#p78975. If you're using your own router, ensure that the modem/router combo (or gateway/hub if you have one) is in bridge mode.

Note that it's not possible to register the exact same FPL account simultaneously on more than one device or sip app. Only one registration per FPL account is allowed at any time. An ATA is one registration. A SIP app is another.

When there are multiple devices/softphones/lines using the same account, only the most recent registration is valid. The previous device will lose registration, and, consequently, incoming calls will not work on it. This is especially important to consider if someone else is using your SIP credentials (username and password) that are found after logging in at https://www.freephoneline.ca/showSipSettings or if you're trying to register your FPL account with a smartphone SIP app or with another device. Registration is required for incoming calls. It is not required for outgoing calls.

A more significant concern, though, is that multiple registration attempts can lead to temporary IP bans. The more devices being used with the exact same FPL account can make the temporary ban happen more quickly. Each time you reboot or restart your ATA or SIP app, it's attempting to register with Freephoneline again. Multiple registration attempts within a short period can result in temporary IP ban. Each time you reboot your ATA (or run a SIP app) it's attempting to register with FPL's proxy server again.

If you've been temporarily IP banned, you can try switching to another proxy server: voip,freephoneline.ca, voip2.freephoneline.ca, or voip4.freephoneline.ca:6060.

voip4.freephoneline.ca:6060 is used to help circumvent buggy SIP ALGs in routers, and it's a good choice to try if you weren't using it before.

Latest Grandstream firmware can be located at http://www.grandstream.com/support/firmware. If yours is outdated, I suggest updating. However, I will not be held responsible for failed firmware updates.

In your Grandstream device, use (if the settings are available):

Primary SIP Server: voip4.freephoneline.ca:6060
NAT Traversal: Keep-Alive
SIP Registration: Yes
SIP Registration Failure Retry Wait Time: 120 seconds
Enable SIP OPTIONS/NOTIFY Keep Alive: OPTIONS
SIP OPTIONS/NOTIFY Keep Alive Interval: 20
SIP REGISTER Contact Header Uses: WAN Address
Register Expiration should be 60 minutes
Use Random SIP Port: Yes
Use Random RTP Port: Yes

This is always proper device reboot order:

A.Turn off modem, router and ATA (or IP Phone or close SIP App).

B. Turn on modem. Wait for modem to be fully up and running.

C.Turn on router.
Wait for modem to be fully up and transmitting data before turning on router.

D. Turn on ATA (or IP Phone or open SIP app) only after the router is fully up and running.

[iceonu]

I factory reset the phone and checked the settings as you provided. It did the trick. Unsure what initially caused it to be unregistered, as I had not implemented any changes and had been working okay prior.

[Liptonbrisk]

Unsure what initially caused it to be unregistered, as I had not implemented any changes and had been working okay prior.

Issues can occur in users' routers, such as NAT corruption, without them doing anything. Consequently, users stating that nothing has changed on their end doesn't necessarily mean very much.
Refer to point D from viewtopic.php?f=8&t=20199#p78976

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, 15, 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_KeepAlivesExpires (SIP OPTIONS Keep Alive Interval) is supposed to be 20 with FPL.








To help ensure this situation doesn’t reoccur. . .

1) Are you using a Telus Wi-fi Hub? If so, ensure its LAN Port that is connected to Ubiquiti router is set to “Bridge”. Contact Telus for assistance.
Otherwise please ensure whatever modem/router combo, hub, or gateway your ISP issued is in bridge mode.

2) In your Unifi Controller (or Cloud Key), navigate to ‘Routing & Firewall’ -> ‘Firewall’ -> ‘Settings’.

Disable "SIP".


3) Refer to point D from viewtopic.php?f=8&t=20199#p78976

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.


In your Unifi Controller (or Cloud Key), navigate to ‘Routing & Firewall’ -> ‘Firewall’ -> ‘Settings

Change
a) udp stream 115 (assured)
b) udp other 15 (unreplied)

The defaults are 180 for Stream and 30 for Other.

(Refer to pages 35 and 36 from https://dl.ui.com/guides/UniFi/UniFi_Co ... _V5_UG.pdf)

So, 15 seconds (UDP Unreplied or Other) < 20 seconds (SIP OPTIONS/NOTIFY Keep Alive Interval: 20) < 115 seconds (UDP Assured or Stream) < 120 seconds (SIP Registration Failure Retry Wait Time)


If you want to keep the router defaults for UDP Stream and UDP Other instead, then you could change, in your IP Phone,
SIP OPTIONS/NOTIFY Keep Alive Interval to 35 seconds
and
SIP Registration Failure Retry Wait Time to 185 seconds
However, in the event of a failed registration, the IP Phone will now take 185 seconds instead of 120 seconds before attempting to register again with FPL's proxy server.
Also, the Keep Alive packets (which is sent to help keep the NAT connection/hole alive) will be sent at 35 second intervals instead of 20. I doubt this would cause a problem for you.



c) click "Apply Changes"


You can also do steps 2 and 3 by SSHing into your router, if you prefer:

configure
set system conntrack modules sip disable
set system conntrack timeout udp stream 115
set system conntrack timeout udp other 15
commit
save
exit

4) Reboot Router. Wait for it to be fully up and running first. Then reboot or power cycle IP phone. This is always proper device reboot order (well, rebooting ISP modem is the first device step, but that shouldn't be necessary here).