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[Solved with Block CID off] One phone number unreachable

PostPosted: 06/15/2020
by houdini
Hi there,

I've been a customer of yours since 2016 and successfully using your service over our landline without serious incident all that time.

But, strangely, I've just encountered a single particular phone number (416-620-6861) that, when I call it from my landline, rings five times and then switches to a rapid busy signal, while if I call it from your desktop app, goes through to the company in question's voice menu system immediately without even a single ring!

Can you think of any conceivable cause that could possibly lead to this strange dichotomy?!

Let me know.

Thanks!

Re: Single particular phone number unreachable through landl

PostPosted: 06/15/2020
by Liptonbrisk
That behaviour (going to busy) can happen when the other party is blocking you.

Is your FPL desktop app using a different FPL account than your home phone?

What brand and model ATA are you using? If you can, check call logs in the device.
Also, login at https://www.freephoneline.ca/doGetCallLogs, and check for abnormal "disconnect reason[s]" for the calls that go to fast busy.
Ensure the "to" field actually shows that you're dialing the correct number.

Re: Single particular phone number unreachable through landl

PostPosted: 06/16/2020
by houdini
It wouldn't be caused by they're blocking me because I'm using the same account to successfully call them through the FPL desktop app.

The successful calls to that number through the FPL desktop app of course show up in my FPL call log, but the attempts through my landline do not.

I'm using a Linksys SPA-2102 which, AFAIK, does not maintain its own call log.

Re: Single particular phone number unreachable through landl

PostPosted: 06/16/2020
by Liptonbrisk
houdini wrote:I'm using a Linksys SPA-2102 which, AFAIK, does not maintain its own call log.


Dial the number, and then check
in your ATA under Voice—>Line—>Last Called Number
to see whether the ATA dialed what you entered

If it didn’t, things to look for include a problem with your ATA’s dial plan or a flaky telephone button (try a different phone).
Try the DTMF test using the 416 number at http://thetestcall.blogspot.com/2012/08 ... mbers.html

You could also try capturing a log of the call to look for anything abnormal:
https://www.cisco.com/c/en/us/support/d ... 08784.html
Syslog executable still seems to be available here: https://community.cisco.com/t5/small-bu ... -p/3293797

If the issue with that specific number is intermittent, then the Auto Attendant may have been temporarily overloaded at the time that you called.

Re: Single particular phone number unreachable through landl

PostPosted: 06/17/2020
by houdini
OK, I've saved a log resulting from attempting to make the call.

What should I look for in it that would provide a clue as to what is going wrong?

(I don't want to post the contents of the file publicly here, in case there might be something in it that would compromise the security of my account or our home network.)

Re: Single particular phone number unreachable through landl

PostPosted: 06/17/2020
by Liptonbrisk
Well, you could edit out your public (WAN) IP address and FPL username, whenever they occur, and post the log.
I would check to ensure the phone number being dialed by the ATA is the real one that you dialed.

And then I would check for 3 digit SIP error codes towards the end of the log, such as SIP error code 480, for example: https://www.ietf.org/rfc/rfc3261.txt (page 188)

480 Temporarily Unavailable

The callee's end system was contacted successfully but the callee is
currently unavailable (for example, is not logged in, logged in but
in a state that precludes communication with the callee, or has
activated the "do not disturb" feature). The response MAY indicate a
better time to call in the Retry-After header field. The user could
also be available elsewhere (unbeknownst to this server). The reason
phrase SHOULD indicate a more precise cause as to why the callee is
unavailable. This value SHOULD be settable by the UA. Status 486
(Busy Here) MAY be used to more precisely indicate a particular
reason for the call failure.

This status is also returned by a redirect or proxy server that
recognizes the user identified by the Request-URI, but does not
currently have a valid forwarding location for that user.



UA = user agent, by the way

480 SIP errors are also caused by capacity issues. FPL offers a maximum of 2 channels (you can have a maximum of two concurrent calls) per account. With some service providers, trying to exceed that 2 channel limit produces that 480 error. Possibly trying to register your FPL account with another device or SIP app simultaneously could cause that 480 SIP error as well. 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 (despite the ATA's registration status). In other words people sharing their FPL accounts or otherwise trying to register their FPL accounts with smartphone SIP apps/desktop apps while simultaneously using the same FPL account on their ATAs are out of luck. http://www.dslreports.com/forum/r28918051- (At that link you'll see the 480 error with FPL).


I'm guessing about the 480 error, but if you see it, I've provided an explanation.

Re: Single particular phone number unreachable through landl

PostPosted: 06/17/2020
by houdini
It's not obvious to me from looking at it.

Here's a link to a minimally censored version of it (to the extent that I can determine anything that warrants censoring!):

*file link removed because I can see your public WAN IP address in the log*--Lipton Brisk

Re: Single particular phone number unreachable through landl

PostPosted: 06/17/2020
by Liptonbrisk
1) Would you be using different DNS servers for your computer than in your ATA? You could try Google DNS (testing) in your ATA to see if that makes a difference.
In your ATA, navigate to Router-->WAN-->Optional Settings-->Primary DNS and Secondary DNS.

2) Also, you're not using "Anonymous" for display name in your ATA, right? If you are, change it to something else for testing purposes.
Navigate to User-->Supplementary Service Subscription-->Block CID Serv/Setting: change to no

3) Lastly, if the problem persists, try changing Voice-->Line-->Proxy to "voip4.freephoneline.ca:6060" without the quotation marks



I see 502 Bad Gateway in your log file. I'm not sure what would be generating that error just for 1 phone number.


https://www.ietf.org/rfc/rfc3261.txt
21.5.3 502 Bad Gateway

The server, while acting as a gateway or proxy, received an invalid
response from the downstream server it accessed in attempting to
fulfill the request

https://www.cisco.com/en/US/tech/tk652/ ... eadfa.html

"Upon receiving this response, the phone notifies the user with fast-busy signal and disconnects the call."

Re: Single particular phone number unreachable through landl

PostPosted: 06/18/2020
by houdini
1) I have our router's local IP address set for both gateway and Primary DNS.

2) As you can see from the fact that at least some instances of "From:" are followed by (the censored version of) my name, I do have my name specified in the 'Display Name' field (in Voice-Line 1 settings). I don't know why some other instances of "From:" are followed by "Anonymous". I'm sure you would be able to speak more to that.

Toggling the 'Block CID Setting' to "no" does fix the problem. Does setting that to "yes" prevent the callee from seeing our caller ID? I didn't choose to do that deliberately and am perfectly fine with them seeing it.

3) Since the latter worked, I haven't tried switching to "voip4.freephoneline.ca:6060" for the proxy server, but I know that's recommended for people who are having problems that might be fixed if their router supported turning SIP ALG off but, like ours, doesn't (as discussed in this post viewtopic.php?f=8&t=19835).

In the past, we've occasionally had problems with people not being able to get through when they try to call our phone. Is that something that would be fixed by using that alternative proxy server?

Re: Single particular phone number unreachable through landl

PostPosted: 06/18/2020
by Liptonbrisk
houdini wrote:2) As you can see from the fact that at least some instances of "From:" are followed by (the censored version of) my name, I do have my name specified in the 'Display Name' field (in Voice-Line 1 settings). I don't know why some other instances of "From:" are followed by "Anonymous". I'm sure you would be able to speak more to that.


The contact header showing "Anonymous" is a result of having Block CID Setting on. The default is no for Block CID Setting.
That setting attempts to use the ATA to block outbound caller ID (CID) on all outgoing calls when enabled.

There's two possible ways *67 (block outgoing CID) works:

1)You can send *67 through the service provider, who then either accepts the code or doesn't do anything. FPL, by default, doesn't do anything. At least that was the case before server migration: viewtopic.php?f=15&t=19702 (I'm just posting that link to show there was a server migration). There have been a few exceptions made for certain individuals, but it's rare.


2) You can have *67 processed by your ATA (or you can have Block CID Setting on), which sends an anonymous call flag in the SIP header along the path to the receiving carrier that, in turn, has to recognize and agree to not show CID. Blocking outbound calls to Telus Mobility numbers work, for example, works when *67 is processed by the ATA. That is, no caller ID is displayed. When *67 (or Block CID setting is on) is parsed by an ATA or IP Phone, my FPL number constantly appears when calling Fongo Mobile numbers. But it's blocked when calling regular Telus mobile numbers, for example. Telus Mobility acknowledges the flag and complies. Fongo Mobile doesn't.

For Cisco ATAs, for some reason, after FPL's server migration earlier this year, users have reported 15 minute call drops and also fast busy when block Block CID Setting is enabled (one user reported the busy signal, but it had slipped by mind by the time you posted). 15 minute calls drops would be due to a lack of (200 OK) acknowledgement (ACK) being received after an INVITE.

As for why having Block CID setting enabled would cause a 502 bad gateway error when only calling one phone number, I have no idea, but the receiving end (the destination you're calling) might reject "Anonymous" when it's processed from the "Contact" header. However, I really have to stress that I really no clue about the 502 bad gateway error when Block CID setting is enabled, and I'm just guessing.




Toggling the 'Block CID Setting' to "no" does fix the problem. Does setting that to "yes" prevent the callee from seeing our caller ID?


That's the intention of that setting. Whether your CID is shown depends on the destination's carrier (method #2 above).



3) Since the latter worked, I haven't tried switching to "voip4.freephoneline.ca:6060" for the proxy server, but I know that's recommended for people who are having problems that might be fixed if their router supported turning SIP ALG off but, like ours, doesn't (as discussed in this post viewtopic.php?f=8&t=19835).

In the past, we've occasionally had problems with people not being able to get through when they try to call our phone. Is that something that would be fixed by using that alternative proxy server?


Using voip4.freephoneline.ca:6060 is useful if you're not sure whether the device (modem/router combo) issued by your ISP has SIP ALG enabled or whether you may also have SIP ALG enabled if you're using your own router. Anyone can use voip4.freephoneline.ca:6060, regardless. SIP ALG (a feature in routers and modem/router combos) monitors traffic on UDP port 5060 and sometimes 5061 and can alter or mangle SIP headers, making SIP signalling in some situations impossible.


i) What brand and model modem are you using?
ii) What brand and model router are you using?

Check NAT Keep alive settings. Linksys/Cisco SPA/PAP users should ensure the following:

A. 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.

Using a high random SIP port may help to bypass SIP ALG, and it also helps to avoid SIP Scanners (or hackers).


B. 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

This helps to ensure RTP audio is being sent to your WAN IP as opposed to your LAN IP.

d) Ensure NAT Keep Alive Intvl is 20 seconds (I think it is already, but I deleted your log file for privacy)


C. 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

D. 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

a) Also navigate to Voice-->Line-->Proxy and Registration-->Register Expires needs to be 3600 seconds (it probably already is set to 3600)

E. Look at point #4 from the post below this one, concerning UDP timeouts.

F. Proper device reboot order is always modem (wait for it to be fully up before turning on your)-->router (ensure Wi-Fi SSIDs are populated first on your devices)-->ATA (wait for router to be fully up and running before turning on ATA).
That's always proper device reboot order.

Registration is a requirement for incoming calls but not for outgoing calls. When you have problems, login at https://www.freephoneline.ca/showSipSettings. SIP status needs to show "connected", and "SIP User Agent" needs to be a device or SIP app you recognize. Ensure that you're not using FPL desktop (or anything) else that uses the same FPL account when trying to receive an incoming call on your ATA.
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.

Re: Single particular phone number unreachable through landl

PostPosted: 06/18/2020
by Liptonbrisk
(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
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.

Router firmware that allows users to adjust Assured and Unreplied UDP timeouts includes

Asuswrt-Merlin
Ubiquiti
Mikrotik
pfSense
Tomato
DD-WRT



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.

Re: [Solved with Block CID off] One phone number unreachable

PostPosted: 06/19/2020
by houdini
So, even though we ourselves don't feel a need to block our caller ID from going out, should I notify the callee company in question that their phone system is preventing potential customers from reaching them if they (the customers) have chosen to do so (i.e. block their caller ID from going out)?

Either way, thanks a lot for solving that problem for us!!

For Cisco ATAs, for some reason, after FPL's server migration earlier this year, users have reported 15 minute call drops and also fast busy when block Block CID Setting is enabled (one user reported the busy signal, but it had slipped by mind by the time you posted). 15 minute calls drops would be due to a lack of (200 OK) acknowledgement (ACK) being received after an INVITE.


We too have experienced call drops after 15 minutes. Is there a user work around for that?

We have a Linksys WRT54G router and a Thomson DCM475 cable modem.

Re: [Solved with Block CID off] One phone number unreachable

PostPosted: 06/19/2020
by Liptonbrisk
houdini wrote:So, even though we ourselves don't feel a need to block our caller ID from going out, should I notify the callee company in question that their phone system is preventing potential customers from reaching them if they (the customers) have chosen to do so (i.e. block their caller ID from going out)?


I wouldn't bother.

As I mentioned before, I'm just guessing. The issue is either caused by enabling Block CID with Cisco ATAs when using FPL only--or it's a conscious decision to block anonymous incoming calls by the party you are calling. If it's an issue specific to using Block CID with Cisco ATAs and only when using FPL, the party you're calling can't doing anything. And if the system they're using is purposely designed to block "Anonymous" voip calls, then they're not going to care either.


We too have experienced call drops after 15 minutes. Is there a user work around for that?


Disable Block CID setting in the ATA (that also caused the 15 minute call drop for some people), which you have already done. And follow the steps to change ATA settings that I mentioned previously.
That is, use voip4.freephoneline.ca:6060 for proxy in the ATA and also follow steps A to F, step by step, from posting.php?mode=quote&f=8&p=77665#pr77661.

We have a Linksys WRT54G router


Official firmware can be found at https://www.linksys.com/ca/support-arti ... Num=148648

Some versions of that router are reported to cause SIP signalling problems: https://www.voip-info.org/linksys-wrt54g/

Switching to DD-WRT firmware would probably be better than using stock Linksys firmware, but I will not be held responsible for failed firmware updates or anything problems arising from switching firmware.

It's easier to use voip4.freephoneline.ca:6060 for proxy in the ATA and to also follow steps A to F, step by step from posting.php?mode=quote&f=8&p=77665#pr77661
You won't be able to do step E using Linksys firmware.

Re: [Solved with Block CID off] One phone number unreachable

PostPosted: 06/19/2020
by houdini
OK, I'll try all that when I find some time and it'll probably be longer till the next time we have a longer than 15 minute call to test it, but in the meantime...

Thanks so very much for all your much valued help!

Re: [Solved with Block CID off] One phone number unreachable

PostPosted: 06/19/2020
by Liptonbrisk
You're welcome! I hope you have a good weekend.