SIP Registration issue

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SIP Registration issue

Postby hieppo » 04/12/2020

I believe there is issue with the FPL SIP registration. I first notice about 5 days ago. I can only make outgoing call but all incoming call would not direct to the ATA adapter. The FPL web control panel shows the registration SIP User Agent and connected so outgoing call was possible. However, incoming call would pause for about 30 seconds before it goes to FPL voicemail.

There are several tests I did and this is what I discovered.

Using voip.freephoneline.ca:
1. Outgoing is successful
2. Incoming failed

Using voip2.freephoneline.ca
1. Outgoing is successful
2. Incoming from outside FPL (other providers and mobile) is successful
3. Incoming from other FPL members majorily failed with exception of some NPA/Exchange test (some registered on voip and some on voip2)

I know there is no support with FPL but this seems to be an FPL issue and not at the user's end. I have been with FPL for over a decade and there has never been any incident like this that took this long to resolve itself.

If anyone else is having this issue, please post and may be moderator may be able to escalate this issue to the FPL support team.
hieppo
Just Passing Thru
 
Posts: 6
Joined: 12/03/2012
SIP Device Name: PAP2
Firmware Version: Sipura 3.1.8
ISP Name: Distributel
Computer OS: Ubuntu
Router: Tomato v1.28

Re: SIP Registration issue

Postby Liptonbrisk » 04/12/2020

Moderators don't really have the capacity to escalate anything since we don't work for Freephoneline. We're volunteers. We can try to submit support tickets or try to send a private message to someone, just like you can, and hope that someone listens to us.

Also, I have no issues registering on any FPL proxy server using any of my FPL accounts.
I have no issues calling or receiving calls between my FPL accounts.
I have no issues calling or receiving calls on any of my FPL accounts, regardless of the proxy server I'm using, and regardless of the source of the call (cell, landline, voip service). I have no issues receiving calls from family members located in another province who are also using FPL (I setup their service and remotely ensure their equipment is configured and running properly on a monthly basis).

1) What brand and model modem are you using?
2) What brand and model router are you using?
3) What brand and model ATA or IP Phone are you using?
4) Are you using some third party PBX service? Freepbx?

The issues you're describing typically involve SIP ALG problems, UDP timeout issues, or a firewall NAT/NAT Keep alive issue. And the issue can also be with someone's equipment on the other end of the call: viewtopic.php?f=8&t=19773#p77307, for example.

The FPL web control panel shows the registration SIP User Agent and connected so outgoing call was possible


Registration is required for incoming calls only--not for outgoing calls, unless that requirement ("registration required for outgoing calls" in Grandstream ATAs, for example) is enabled in a device. 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 and incoming calls will not work on it.
Also, note that too many registration attempts within a short period can result in a temporary IP ban with the server you're attempting to register with.

You should also be checking registration status in your SIP device(s) in addition to the connection status from your FPL account(s).
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: 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: SIP Registration issue

Postby hieppo » 04/12/2020

I am on Bell Fibe GPON using their HH3000 but I bridge to my pfsense router and it is PPPoe to Bell to obtain the external IP.
I am using Asterisk PBX 16.5.1. Internally, I am using both Sipura ATA and Nortel SIP phones to register on my Asterisk.

This setup has been working for a while now without any issues.

I started noticing hiccups several days ago. The most important was my hour call were getting drop about 4 times over the hour period. Afterwards, i notice I was not able to call my parent's number who is register on voip.freephoneline with Sipura ATA and my b-in-law register on the same server with OBiHai201. At this time, I was also registered on voip.freephoneline. I also tested incoming using my mobile to call my number and it failed.

I read about voip2 on the forum and changed the proxy server and it solved my problems of incoming call but I was still not able to call my father or b-in-law who was on voip.

All settings have not been changed over months. The only change was myself on Asterisk to swtich proxy server to voip2.

On another note, I just registered a free Fongo account and used the app to call my parent's house and it worked perfectly fine.
hieppo
Just Passing Thru
 
Posts: 6
Joined: 12/03/2012
SIP Device Name: PAP2
Firmware Version: Sipura 3.1.8
ISP Name: Distributel
Computer OS: Ubuntu
Router: Tomato v1.28

Re: SIP Registration issue

Postby Liptonbrisk » 04/12/2020

hieppo wrote:I am on Bell Fibe GPON using their HH3000 but I bridge to my pfsense router and it is PPPoe to Bell to obtain the external IP.
I am using Asterisk PBX 16.5.1. Internally, I am using both Sipura ATA and Nortel SIP phones to register on my Asterisk.


1) Using *67 or having Block CID Serv enabled may cause issues for outgoing calls (and 15 minute call drops with PAP/SPA ATAs) after server migration
viewtopic.php?f=8&t=19749&p=77088&p77088#p77088
viewtopic.php?f=8&t=19749&p=77088&p77088#p77432
viewtopic.php?f=8&t=19806#p77432 (main thread)

A number of asterisk and Freepbx users have been reporting problems since server migration: viewtopic.php?f=15&t=19702 (I'm just posting this to show there's been a server migration and that people using IP addresses instead of proxy server names may encounter problems).

2) Some Asterisk/Freepbx users have also resolved their issues:
viewtopic.php?f=8&t=19803&hilit=asterisk&start=25#p77295 (canreinvite=no)
viewtopic.php?f=8&t=19750&p=77117&hilit=asterisk#p77118 (caller id)
viewtopic.php?f=8&t=19740&hilit=asterisk#p77062 (caller id)

I'm not going to be able to help with Asterisk or Freepbx. Too time consuming for me since I'm not using them and don't feel like setting up servers.
But canreinvite=no seems to be important as are Caller ID settings.


Afterwards, i notice I was not able to call my parent's number who is register on voip.freephoneline with Sipura ATA


It would be useful to know the specific model numbers, but visit http://forums.redflagdeals.com/freephon ... #p26808549 and http://forums.redflagdeals.com/anyone-h ... #p26234003

In your PAP2T, Navigate to the SIP tab-->NAT Support Parameters, and make sure that the following settings are enabled:
(Thanks to Mango)
a)Handle VIA received-->yes
b)Handle VIA rport-->yes
c)Substitute VIA Addr-->yes



I'm not permitted to post more than 10 urls per post, so I'll continue in the next post.
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: 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: SIP Registration issue

Postby Liptonbrisk » 04/12/2020

(continued)

Keep-Alive NAT settings that are found in the respective setup PDF guides are also very important.
viewtopic.php?f=15&t=16294
viewtopic.php?f=15&t=16206#p64088



and my b-in-law register on the same server with OBiHai201.


There is no OBi201, by the way.

I'm using an OBi202, and so are my family members in another province. We have no issues. The PDF guide located at download/file.php?id=2065 should be used fully (a lot of people use Obitalk with Freephoneline's preconfigured profile, which can lead to problems). After using the PDF guide fully, look at pages 42 to 44. Similarly, visit http://forums.redflagdeals.com/newegg-o ... #p26724395, and follow the steps, step by step, located at the bottom of the post.
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: SIP Registration issue

Postby hieppo » 04/12/2020

Thanks Liptonbrisk for all your info. I browsed to each link to see the issues. Unfortunately, none of those links are related to my issue at this time.

The drop calls was only happening that one day and have not seen it since. It was just a signal that got me testing the incoming issue. But the disconnect call has not happen since.

Seems most issue is with SIP ALG. pfsense does not use SIP ALG. I will have to check my parent's and b-in-law and see if I can have a workaround to see if it will work.

B-in-law call to my parent's number and vice versa does not work. Both register on voip. I called from my number to my friend's number who are both registered on voip2 and failed to call.

However, from Fongo to FPL can call without issue whether voip or voip2.
hieppo
Just Passing Thru
 
Posts: 6
Joined: 12/03/2012
SIP Device Name: PAP2
Firmware Version: Sipura 3.1.8
ISP Name: Distributel
Computer OS: Ubuntu
Router: Tomato v1.28

Re: SIP Registration issue

Postby Liptonbrisk » 04/12/2020

hieppo wrote:
Seems most issue is with SIP ALG.


With DSL hubs or modem/router combos, if the user's ATA contains a NAT firewall (such as an OBi202), I may just get the user to perform PPPoE in the ATA. That rules out any ISP, device issued, buggy router problem since the ISP's router is being bypassed at that point. Modem/router combos (gateways or hubs) issued by ISPs often have SIP ALG enabled (then just disable it) or SIP ALG enabled and hidden with no way for the customer to disable SIP ALG. If the user has a router, ensure the ISP's modem/router combo is in bridge mode. Then check the user's router for SIP ALG and disable it if the user is having issues.



pfsense does not use SIP ALG



Unless you installed siproxd

And there's also UDP timeouts for you to consider: https://forums.redflagdeals.com/freepho ... #p30454417. I encourage you to read that post.

Moreover, the people you're calling might have SIP ALG enabled: viewtopic.php?f=8&t=19773#p77307. That user is trying to call his parents, who also use FPL, and can't (presumably, after server migration). Then he disables SIP ALG in their router. Problem solved.

I called from my number to my friend's number who are both registered on voip2 and failed to call.


Again, the problem could be on their end (in particular if they have SIP ALG enabled), unless you have Block CID Serv enabled in your ATA.

*67 may be causing calls to drop after 15 minutes when using Linksys/Cisco ATAs when calling mobile numbers.
Those two issues have been reported after server migration.


There's also this new caller ID number format issue involving Asterisk and FPL, which would make outgoing calls impossible, but if that doesn't happen with all your outgoing calls, it probably doesn't apply to you: viewtopic.php?f=8&t=19750&p=77117&hilit=asterisk#p77118.

On their end, check NAT Keep alive settings. Linksys/Cisco SPA/PAP users should check for the following:

1) Specify a high random sip port in your ATA between 30000 and 60000.
Navigate to Line 1 (or whatever you're using for FPL)-->SIP settings, change SIP Port to a random number between 30000 and 60000.
Do not use the same random SIP port for any other Line. Always choose a different random local SIP port for each FPL account you're using.

2. In the ATA, navigate to Voice-->SIP tab-->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) Ensure NAT Keep Alive Intvl is 20

3. Navigate to Voice-->Line (whichever you use for FPL)-->NAT settings
a) NAT Mapping Enable should be yes
b) NAT Keep Alive Enable should be yes

4. This is not related to your problem, but a lot of guides don't have this setting specified:
Navigate to Voice-->SIP-->SIP Timer Values (sec)
Reg Retry Intvl should be 120 seconds


For OBi2xx ATAs (they should really be using this PDF guide fully instead of using obitalk: download/file.php?id=2065),

If you used the Obitalk web portal (http://www.obitalk.com) to configure your ATA, keep in mind that you must continue using it to configure your ATA. Otherwise whatever settings you change will eventually be overwritten by what you previously entered at obitalk.com anyway. If you wish to disable this behaviour, dial ***1. Enter that IP address into a web browser. Navigate to System Management-->OBiTalk Provisioning-->select Disabled for the method. Save. Reboot ATA. Now obitalk.com won't overwrite whatever changes you make via the device's interface (via IP address).

Pick one method (obitalk.com) or the other (IP address of device) for changing device settings. But do not use both methods.

(In Obitalk.com, you will need to enable and enter expert settings to do the following, if you want to use Obitalk.com. You do this by selecting Edit Profile-->Advanced Options-->check Enable OBi Expert Entry from Dashboard-->submit))

Keep in mind too, that if you're using the Obitalk.com web portal, after you submit a new setting, it takes several minutes before Obitalk.com pushes the changes you've made to your ATA. Your ATA should reboot automatically after the changes are submitted.

A. In your Obihai ATA or at Obitalk.com (whichever method you normally use; don't use both methods), navigate to Voice Services-->SP(FPL) Service-->X_UserAgentPort. X_UserAgentPort should be a random port number between 30000 and 60000. Just pick a port number in that range. Change to a new port number in that range. Click the “submit” button, and reboot the ATA. (If you use Obitalk.com to change settings, you will need to continue using Obitalk.com).

B. Navigate to Service Providers-->ITSP Profile (FPL)-->SIP
i) ensure X_DiscoverPublicAddress is enabled (it is by default)
ii) enable X_UsePublicAddressInVia (it's not by default)
You will need to uncheck default, device default, and Obitalk settings boxes. Then check the box to enable the feature.

C. Navigate to Voice Services-->SP(FPL) Service-->
i) Check or enable X_KeepAliveEnable.
ii) Change X_KeepAliveExpires to 20.
iii) Select “notify” from the dropdown box for X_KeepAliveMsgType


And for people in general having problems with incoming calls, I would ensure that they're using voip.freephoneline.ca:6060 for the proxy server to help circumvent faulty SIP ALG features in routers.

(save settings)

Afterwards, these people should rebooting their modems–>router (wait for it to be fully up and transmitting data; wait for Wi-Fi SSIDs to populate first)–>ATA (in that specific order). That's always proper device reboot order.
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: SIP Registration issue

Postby Liptonbrisk » 04/12/2020

Concerning UDP timeouts, refer to #4 below (for other users that you're calling). For pfsense, I provided a link in the previous post.
Mostly I'm posting this for things to consider when calling other users (problems on their end, in particular, #2 and #4 below).


Here's some general information on routers, since many people don't know what features to look for when buying routers for SIP services, such as FPL, in the event you're also interested.


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

Re: SIP Registration issue

Postby Liptonbrisk » 04/12/2020

Also, to test, I've used one FPL account configured on a SIP app called Groundwire connected via cellular data to call another FPL account configured on an Obihai ATA. So the two FPL accounts are connecting on two separate networks. I tested on all 3 proxy servers. I am able to also call and receive calls from family members using FPL.
All calls work fine, incoming and outgoing. Incoming audio and outgoing audio work. No problems.
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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