FPL with HT-701 (Firmware 1.0.8.2)

[Liptonbrisk]

So far so good. SIP OPTIONS Keep Alive Interval seems to have changed to a default of 30, whereas my guide calls out 20. That's the only difference I've noticed so far. Will monitor a little more before updating the guide.

It should be 20.

https://support.freephoneline.ca/hc/en- ... redentials
"Keep Alive Interval: 20 seconds"

Your guide is correct.

[Melchizedek]

Hi, thank you very much for your guide, I have recently acquired an HT702, updated its firmware, reset it to default and followed this guide successfully.

I have two questions:

1) Can you tell me how I could specify, in the Dial Plan, that dialing 411 should get me to 111-111-1111, and that dialing 811 should get me to 222-222-2222?
2) I will need to configure a second HT-701/HT-702 device, but this time with voip.ms. It looks like the voip.ms configuration guide is a little dated (based on firmware 1.0.0.18). Should I still follow the outdated voip.ms guide, or follow this FPL guide instead and replace the credentials?

Thanks!

[Liptonbrisk]

1) Can you tell me how I could specify, in the Dial Plan, that dialing 411 should get me to 111-111-1111, and that dialing 811 should get me to 222-222-2222?

You're in Quebec, so you don't need to do that: https://support.fongo.com/hc/en-us/arti ... 1-811-911-.
All you need in your dial plan is [2-9]11

But <811=2222222222> does what you're asking.

I will need to configure a second HT-701/HT-702 device, but this time with voip.ms.

We don't support other VoIP services here. Please contact voip.ms.

[Melchizedek]

I have configured both ATAs with FPL, but even though both my relatives use the same router and the same ISP, the one with the HT-702 experiences some registration issues. His device randomly loses registration, even though it is configured according to this guide. Do you have any advice?

[Liptonbrisk]

I have configured both ATAs with FPL, but even though both my relatives use the same router and the same ISP

It doesn't matter if it's the same ISP unless both ATAs are located at the exact same address and attached to the same router. Intermittent congestion and connectivity issues can occur regardless of whether the same ISP is used at two different locations.

If both ATAs are at the same physical location, do not use the same Primary SIP Server in both ATAs, unless you have no other choice (i.e., unless you're forced to use voip4.freephoneline.ca:6060 due to SIP ALG problems). Local SIP and RTP ports should not be the same (if set to random the odds are unlikely).

Please read the preamble (the first four pages) of this PDF guide: http://forum.fongo.com/viewtopic.php?f= ... 805#p73839 (especially point #6 on page 3). I realize the guide is for a different ATA, but the preamble, generally, applies to everyone. Additionally, ensure that QoS is properly enabled for ATAs in the router being used (refer to the router manual, and also refer to page 45 in the same PDF guide).


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. 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. If you simply want to make outgoing calls using your FPL number, configure, but don't register the account, on the SIP app being used. This is also important to consider if you're using Freephoneline's desktop application (don't have it running while using your ATA with the same FPL account). Additionally, keep in mind that if someone else is also attempting to register the same SIP credentials on another device where you live, too many registration attempts can result in a temporary IP ban.

If your ATA makes more than 5 registration attempts in 5 minutes, you may end up being temporarily IP banned by the specific FPL server the ATA was sending registration requests to. If you're temporarily IP banned, you could then try switching ProxyServer (refer to pages 14, 20, and 21) to a different FPL server than the one you were previously using (voip.freephoneline.ca, voip2.freephoneline.ca, or voip4.freephoneline.ca:6060), unless you need to use voip4.freephoneline.ca:6060 because you have SIP ALG forced on in your router. The purpose of voip4.freephoneline.ca:6060 is to circumvent SIP ALG features in routers.

From https://community.freepbx.org/t/trunk-s ... ca/22479/8
"As May 2013, our servers will rate limit REGISTER requests to a maximum of 10 requests per 5 minutes. Each authentication round usually consumes 2 requests (digest auth), so it is a fair number given our guidelines. Also, it does not affect INVITES (which are also authenticated)...

This rate limit is applied per IP address as our service is tailored to residential Canadian users (ADSL/Cable)."


Each time you reboot an ATA, it attempts to register with FPL.
If you're temporarily IP banned, you could then try switching Primary SIP Server to a different FPL server than the one you were previously using (voip.freephoneline.ca, voip2.freephoneline.ca, or voip4.freephoneline.ca:6060), unless you need to use voip4.freephoneline.ca:6060 because you have SIP ALG forced on in your router. The purpose of voip4.freephoneline.ca:6060 is to circumvent SIP ALG features in routers. Or you can disable FPL SIP registration in all devices (turn off your ATA) and wait several hours until the temporary ban clears.

[Liptonbrisk]

. Sometimes people would call me on my mobile saying that they would get a message “User unavailable” (something like that!) and get a voicemail when calling my home phone. While I am sitting by the phone and it does not ring. Then I would reboot my ATA. When Call Logs are checked, you will see this “Destination out of order”:

If rebooting helps with local SIP port set to random, a corrupted NAT association likely developed between your router and your ATA.
The problem can arise due to your router's UDP timeout values (point #3 below).

Your ATA needs to be registered with FPL in order to receive incoming calls. The next time you encounter this issue, check registration in the ATA and also after logging in at https://www.freephoneline.ca/showSipSettings. SIP status needs to show "connected." Also, the SIP user agent needs to show your ATA.

This issue comes and goes, so I’m not sure it this is related to my old firmware OR old settings OR this is normal and everyone encounter this from time to time?

It's not normal behaviour for a properly configured ATA used in conjunction with a good router (as defined in the second post below), unless your internet service is dropping intermittently.

1. Use this PDF guide to ensure your ATA is configured properly: http://forum.fongo.com/viewtopic.php?f= ... 839#p74000.
Check all settings, especially

a. Enable SIP Options Keep Alive: Yes
b. SIP OPTIONS Keep Alive Interval: 20
c. SIP REGISTER Contact Header Use: WAN Address
d. Use Random SIP Port: Yes
e. Register Expiration: 60 minutes (default)
f. SIP Registration Failure Retry Wait Time: 120 seconds


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

By the way, modem/router combos or gateways that ISPs issue are predominately bad choices for use with SIP services. Especially refer to #D in the second post below


3. 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) < SIP OPTIONS 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 (SIP Options) 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, SIP OPTIONS Keep Alive Interval is supposed to be 20 with FPL.



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



4. https://community.freepbx.org/t/trunk-s ... ca/22479/8
"As May 2013, our servers will rate limit REGISTER requests to a maximum of 10 requests per 5 minutes. Each authentication round usually consumes 2 requests (digest auth), so it is a fair number given our guidelines. Also, it does not affect INVITES (which are also authenticated)…

This rate limit is applied per IP address as our service is tailored to residential Canadian users (ADSL/Cable)."

That's why the registration interval of 3600 seconds and failed retry timer of 120s in the ATA are important.

If the ATA loses registration for any reason, incoming calls won't work on it. 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. 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. If you simply want to make outgoing calls using your FPL number, configure, but don't register the account, on the SIP app being used. This is also important to consider if you're using Freephoneline's desktop application (don't have it running while using your ATA with the same FPL account). Additionally, keep in mind that if someone else is also attempting to register the same SIP credentials on another device where you live, too many registration attempts can result in a temporary IP ban. Always check registration status in the IP Phone and also your SIP status after logging in at https://www.freephoneline.ca/showSipSettings. If you see a device listed under SIP User Agent that you don't recognize, you've either been hacked or someone else is using your Freephoneline SIP username and SIP Password.

Always check registration status both in your ATA first and at https://www.freephoneline.ca/showSipSettings if incoming calls aren't being received.

[Liptonbrisk]

I opened this thread so that mh1983 could have access to it if need be.

By the way, I recommend SIP Passthrough in the latest official Asus firmware be set to Disabled. Something has changed that requires it to be disabled in some cases, and enabling it is a potential security risk.

In Merlin, just don't have SIP Passthrough set to "+NAT Helper". NAT Helper is the ALG.

Keep SIP ALG off. Thanks.