*67 Anonymous call/CID Block Serv fast busy/15 minute drop

[Cyber]

@Liptonbrisk,

I now suffer from a new issue.

Outgoing calls disconnect after 15 minutes when calling cellulars phones or landlines.

No dropping calls between freephoneline SIP calls.

Incoming calls are not affected

[Liptonbrisk]

Outgoing calls disconnect after 15 minutes when calling cellulars phones or landlines.

All outging calls to cells?

And at exactly 15 minutes?

These problems are really hard to troubleshoot without SIP traces/logs.

Okay, I'm going to have to test this, I guess, but I'm not thrilled with the prospect of tying up my cellphone line.
I'll let you know.




1) One possibility is an ACK response isn't being received and, in turn, the call drops. Typically that's SIP ALG related, and usually only after 32 seconds and not after 15 minutes with FPL in the past.

Another possibility is the issue is on FPL's end (I hope I don't encounter the same problem when I test), which would be very annoying to troubleshoot. I encountered a similar issue once with another provider that only uses IP for authentication, but that's another discussion.

If you have SIP Passthrough (this is SIP ALG in Asus routers) set to "Enabled + NAT Helper", this shouldn't be happening.


When calls drop after 32 seconds, the problem is the ACK (acknowledgement) response isn't being received after the 200 OK status.



200 OK is sent to your ATA by Freephoneline's server. Your ATA is supposed to send ACK (acknowledge) back. If the ACK response is not received by Freephoneliine's server, then
Freephoneline's server eventually stops waiting for ACK and replies with BYE (typically at 32 seconds), ending the call. SIP signalling can be completely broken by faulty SIP ALG features in routers or
modem/router combos.

With some providers, the renegotation can repeat at different intervals. I used to have a problem with one provider around the 14 minute mark. But that was fixed after adjusting the SIP ALG setting in my router.



2) Another possibility is a lack of QoS being set in your router. This should be done regardless.
Click https://www.asus.com/US/support/FAQ/1008717.
Assign your VoIP device/ATA highest priority and give everything else on your Local Area Network a lower priority than your VoIP device.

In order for Bandwith Monitor (found under Adaptive QoS in your router) to work properly, first, you must turn all devices in your household that connect to the internet (smartphones, computers, smart thermostats, hubs, console gaming systems, roku/apple tv, etc.) so that they are listed.

Drag and drop "Highest" onto your VoIP device (ATA). Do not assign "highest" to anything else. Assign a lower priority to everything else (it's your choice for what priority to use for each device--but don't use "highest" on anything other than your VoIP device). Click "apply".

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 (randomly) have access to enough bandwidth to maintain a call 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 calls would randomly drop due to ping spikes (for the ATA).


3) Another possibility is you're dealing with congestion, especially during prime time (8p.m. to 11 p.m., especially on Sundays). That's an ISP issue (possibly oversold its service in your area). With Rogers or a cable ISP, you could very well be dealing with local node congestion.

Try running http://vac.visualware.com/ at 8p.m. (especially on a Sunday).

After visiting the link, choose a test location that’s closest to server (Freephoneline’s servers, at this time, are in southern Ontario) you’re using. A MOS score below 4.0 is bad news. It means call quality will not be good. The advanced (+) tab provides interesting info.

You should also try this winmtr test: https://forums.redflagdeals.com/newegg- ... #p27515963
.
WinMTR can be found here: https://sourceforge.net/projects/winmtr/

If you're getting bad jitter, calls will sound choppy. With enough packet loss, the calls can drop.


If the problem only occurs during prime time (as opposed to weekday mornings), then I would probably start thinking your ISP is to blame.

4) Another possibility is that your ISP uses poor routing tables to FPL's servers. This usually causes pings to increase. Large ISPs typically won’t do anything if you complain. Smaller ones might.

No dropping calls between freephoneline SIP calls.

Then I doubt points 3 and 4 are the issue, unless the calls drop randomly.

Anyway, I'll try a test call.

[Liptonbrisk]

Outgoing calls disconnect after 15 minutes when calling cellulars phones or landlines.

Okay, I'm using voip4.freephoneline.ca:6060, since you said you were as well, to place an outgoing call to a regular cell number. I'm at the 30 minute mark without the call dropping. I'm going to hang up.
I would go over the 4 points I listed earlier. If you're encountering random call drops, it's either a QoS issue on your LAN (local area network) or an ISP (possibly a cable signal strength or local node congestion if you're on cable internet) issue. ISP issues are extremely annoying to troubleshoot

I once helped a friend who was having problem with Rogers cable internet. Signal strength to the modem seemed fine, but random packet loss was occuring. Calls would drop randomly. And she could also see the problem when playing online games. Suddenly a character would stop moving for a few seconds. Eventually, a Rogers truck had to roll out down the street to change something.

[Cyber]

Before the migration, I never had all these issues.

All my outgoings calls to cell phones (Koodo, Telus) or landlines drops at 15 minutes and 04 seconds.

My router Asuswrt-Merlin (RT-AC68U) SIP Passthrough is set on "Enabled + NAT Helper". I tested with SIP Passthrough on "disabled". Same dropping calls issue.

I changed my NAT Keep Alive Msg for $PING and same issue. This is not a QoS issue. I have 400mbps cable with no bandwidth use on it.

I did lot of test without success. When I call my second FPL line or my friends FPL line, there is no issue.

[Liptonbrisk]

Before the migration, I never had all these issues.

All my outgoings calls to cell phones (Koodo, Telus) or landlines drops at 15 minutes and 04 seconds.

Possibly ACK isn't being received then. I'm not sure why that would be.

You'd need to enable logs, make a call to a cellphone, and then check the log for issues: https://www.cisco.com/c/en/us/support/d ... pters.html.

This starts becoming a lot of work that I'm, frankly, not being paid for, and unfortunately, I'm busy.

Maybe collect the log, zip it up with a password, upload it somewhere, and give FPL support the link to file along with the password: https://support.fongo.com/hc/en-us/requests/new.
Submit a ticket. I realize they don't normally offer free technical support for what they feel are device configuration issues, but maybe they'll make an exception since this problem is happening for you after server migration.
Let them know who the carrier is of the number you're dialing (for example Bell, Telus, Rogers), and also give them the phone number you're dialing in case this is a carrier issue.
If no one responds to your support ticket after 48 business hours, provide the ticket number in a private message to Fongo Support: http://forum.fongo.com/ucp.php?i=pm&mode=compose&u=7852.
I don't know whether they'll agree to look into it, but they might.

Keep in mind, Fongo's status website still indicates "degraded performance": https://status.fongo.com/.

I would suggest seeing if you have the exact same problem with Freephoneline's desktop application for Windows first: login at https://www.freephoneline.ca/downloadDesktopApplication.
If you have problems getting it installed, visit http://forum.fongo.com/viewtopic.php?f= ... 63&p=74810.

My router Asuswrt-Merlin (RT-AC68U) SIP Passthrough is set on "Enabled + NAT Helper". I tested with SIP Passthrough on "disabled". Same dropping calls issue.

Keep it on "Enabled + NAT Helper". That's what it should be set to.

I changed my NAT Keep Alive Msg for $PING and same issue.

SPA series ATAs should use $NOTIFY for FPL

This is not a QoS issue. I have 400mbps cable with no bandwidth use on it.

Even with a 1Gpbs service, problems can still easily occur, especially with cable services due to upload bandwidth being far less. But you should know what's going on on your LAN, so you're probably right.
Regardless, proper VoIP practice enables that feature and configures it properly. Then that would be one less potential headache to deal with.

[Cyber]

2020-01-20 - 10:48 (EST) ISSUE SOLVED


After more than 4 hours of calls test, I found what cause the outgoing call drop.

Before the migration, blocked CID calls was showing "ANONYMOUS". After the migration, blocked CID calls show "unavailable".

The issue is with the Caller ID. I turned off Block CID Serv and no more call drop at 15 minutes 04 seconds

[Liptonbrisk]

Before the migration, blocked CID calls was showing "ANONYMOUS". After the migration, blocked CID calls show "unavailable".

The issue is with the Caller ID. I turned off Block CID Serv and no more call drop at 15 minutes 04 seconds

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. There have been a few exceptions made for certain individuals, but it's rare. My FPL account doesn't do anything for *67 after server migration.

2) You can have *67 processed by your ATA, which sends an anonymous call flag in the SIP header along the path to the recieving carrier that, in turn, has to recognize and agree to not show CID. *67 (outgoing CID block) calls to Telus mobile numbers work when *67 is processed by the ATA.

In this case you're doing #2. When *67 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.

If my FPL account accepted the *67 code (when sending *67 through FPL and not having the star code processed by my ATA instead) and agreed to block outgoing CID, my FPL number would not appear when calling Fongo Mobile numbers. Refer to point #1.

Regardless, it would be pretty strange for a check to occur at the 15 minute 04s mark for an anonymous call flag in a SIP header and then suddenly drop the call. In fact, I would venture to say what you're experiencing is coincidental. After testing with *67 (method #2), which displays "No Caller ID" for me on the receiving end, my call hasn't dropped after 17 minutes.

[Cyber]

With all the tests we did, I can assure you that my outgoing drop call issue is caused by Block CID Serv.

Turning OFF "Block CID Serv" in the ATA fix the issue.

Turning ON "Block CID Serv" cause the drop calls when I call mobile phones.


Probably a SIP header bug for outgoing calls toward mobile phones.

[Liptonbrisk]

With all the tests we did, I can assure you that my outgoing drop call issue is caused by Block CID Serv.

Turning OFF "Block CID Serv" in the ATA fix the issue.

Turning ON "Block CID Serv" cause the drop calls when I call mobile phones.


Probably a SIP header bug for outgoing calls toward mobile phones.

I mean no offense when I write the following: honestly, that situation doesn't make any sense, logically. If either the receiving carrier or FPL is going to drop the call for not having outgoing CID, they're going to drop it immediately. The check is not going to suddenly occur and drop the call after 15 minutes.

If you rebooted your ATA or router after making that change, it's possible an odd NAT issue that developed between the router and ATA was reset.

After testing with *67 (method #2), which displays "No Caller ID" for me on the receiving end to a regular cellphone (non-VoIP) number, my call didn't drop after 17 minutes. Anyway, hopefully, the service continues to work without issues for you and your friends.

[Cyber]

The bug seem to be the "Block CID Serv" feature with the SPA112.


TEST #1 :

Block CID Serv set to "NO"
Display Name set to "Anonymous"

I called my Koodo mobile phone. Caller ID and call log on my mobile phone show "Private Call".
No disconnect after 15 minutes.


TEST #2 :

Block CID Serv set to "YES"
Display Name set to "Anonymous"

I called my Koodo mobile phone. Caller ID on my mobile phone show "Private Call" and call log show "unavailable".
The call drop after 15 minutes.


---------------------------------------


The issue is related with the "Block CID Serv" feature and "unavailable" Caller ID in my mobile phone call log.

[Liptonbrisk]

Display Name set to "Anonymous"

Do you mean that you're actually entering "Anonymous" under Voice-->Line-->Subscriber Information-->Display Name?

Okay, I don't have your ATA to test, but the call dropping after 15 minutes still doesn't make any sense for simply having Block CID Serv enabled.

[Cyber]

Do you mean that you're actually entering "Anonymous" under Voice-->Line-->Subscriber Information-->Display Name?

Yes.

Okay, I don't have your ATA to test, but the call dropping after 15 minutes still doesn't make any sense for simply having Block CID Serv enabled.

I know that doesn't make any sense. Maybe a bug in the latest SPA112 (1.4.1 SR5) firmware.

[sic698]

Hi,

Since this afternoon I'm getting a fast busy when making outgoing calls using my freephoneline account and ata voip adapter?

Incomming calls work ?

Are you experiencing network issues on your end FPL ?

Can you please look into this ?

[Liptonbrisk]

I am not experiencing issues completing calls to numbers I normally dial.

However, Fongo's status page indicates "degraded performance" at https://status.fongo.com/.
There's been a server migration: http://forum.fongo.com/viewtopic.php?f=15&t=19702.

1. What brand and model modem are you using?

2. What brand and model router are you using?

3. What brand and model ATA are you using?

4. When you log in at https://www.freephoneline.ca/showSipSettings

a. what does your SIP status indicate?
b. does your SIP User Agent indicate something you recognize?

5. When you check your call logs
a. at https://www.freephoneline.ca/doGetCallLogs (select the current month), do you notice anything abnormal?
b. in your ATA, do you see a SIP error code?

6. Does the problem only occur when you're calling a specific phone number, or does the problem occur when you're dialing all phone numbers, including, Shoppers Drug Mart, for example?
I have no issues calling 519-432-1809.


I would suggest seeing if you have the exact same problem with Freephoneline's desktop application for Windows first after unplugging power from your ATA. Ensure your FPL account is unregistered on all devices first. Login at https://www.freephoneline.ca/downloadDesktopApplication.
If you have problems getting it installed, visit http://forum.fongo.com/viewtopic.php?f= ... 63&p=74810.

[sic698]

1. What brand and model modem are you using? Thomson 475 cable modem

2. What brand and model router are you using? Mikrotik

3. What brand and model ATA are you using? Linksys/SPA1001

4. When you log in at https://www.freephoneline.ca/showSipSettings Connected

a. what does your SIP status indicate? Connected
b. does your SIP User Agent indicate something you recognize? Linksys/SPA1001

5. When you check your call logs
a. at https://www.freephoneline.ca/doGetCallLogs (select the current month), do you notice anything abnormal? No
b. in your ATA, do you see a SIP error code? No

6. Does the problem only occur when you're calling a specific phone number, or does the problem occur when you're dialing all phone numbers, including, Shoppers Drug Mart, for example?
Problem occurs when dialing all phone numbers, any local number I dial goes to fast busy incoming calls work just fine.

[Liptonbrisk]

What brand and model router are you using? Mikrotik

1. Disable SIP ALG in it: https://www.maxo.com.au/support/trouble ... ik-routers.

or

Mikrotik SIP ALG function is called SIP Helper and found under /IP>Firewall>Service ports
To disable it, run this command from winbox: https://download2.mikrotik.com/routeros ... winbox.exe
/ip firewall service-port disable sip"

https://wiki.mikrotik.com/wiki/Manual:W ... ing_Winbox


Alternatively, register using voip4.freephoneline.ca:6060 instead of voip.freephoneline.ca:5060 or voip2.freephoneline:5060. It doesn't matter that you're not using a Rogers Hitron gateway or modem/router combo; voip4.freephoneline.ca:6060 server works for everyone.
The purpose of voip4.freephoneline.ca:6060 is to cirvcument buggy SIP ALG features in routers.

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).

2. I would suggest seeing if you have the exact same problem with Freephoneline's desktop application for Windows after unplugging power from your ATA. Ensure your FPL account is unregistered on all devices first. Login at https://www.freephoneline.ca/downloadDesktopApplication.
If you have problems getting it installed, visit http://forum.fongo.com/viewtopic.php?f= ... 810#p74810.

I would try testing with SIP ALG disabled in your router. I would also try testing with it enabled to see if it makes a difference with the desktop application.

I would test the desktop application first. If the problem also exists using FPL's desktop app, then there's probably an issue with your account or a carrier issue that I haven't encountered. In that case, no amount of troubleshooting is going to help. You would need to submit an online ticket and ask for someone to check your account: https://support.fongo.com/hc/en-us/requests/new.
But technical support for device configuration issues or issues involving NAT corruption on the user's end is not free (which is why I would test the desktop application first).
If no one responds to your support ticket after 48 business hours, provide the ticket number in a private message to Fongo Support: http://forum.fongo.com/ucp.php?i=pm&mode=compose&u=7852.

Linksys/SPA1001

3. Login to your SPA1001. Selected "advanced" on the right.

4. In your SPA1001, navigate to Line 1 (or whatever line you're using for FPL)-->SIP settings, change SIP Port to a random number between 30000 and 60000. Do this for security reasons.SIP scanners always probe UDP 5060 and 5061 ports first. And changing the local SIP port can also help to reset a corrupted NAT association between your router and ATA.

5. In your SPA1001, navigate to the SIP-->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

Save settings.

Step 5 is extremely important since it helps to ensure RTP audio packets are being directed to your WAN IP address instead of somewhere completely useless, such as a LAN IP. This step should be done, regardless. If incoming calls work for two-way audio, step 5 is probably not related to your specific problem, but this step should be done regardless.

6. Navigate to SIP-->SIP Timer Values (sec)-->Reg Retry Intvl should be 120 seconds
(This is just a failed registration retry timer that won't help with your problem, but that's what the setting should be for Freephoneline. It wasn't included in older setup guides).

7. In your ATA, under Line (whichever one you use for FPL), double check your dial plan for errors. FPL's recommended Linksys dialplan can be found after logging in at https://www.freephoneline.ca/showSipSettings.

8. I would reboot the router at this point. Wait for it to be fully up and running first. Then reboot the ATA. That's always proper device reboot order.
Also, you can try the section at this link called "Workaround for Stale NAT Mappings": https://www.maxo.com.au/support/trouble ... ik-routers.

b. in your ATA, do you see a SIP error code? No

Problem occurs when dialing all phone numbers, any local number I dial goes to fast busy incoming calls work just fine.

Try dialing Shoppers Drug Mart: 519-432-1809. You should hear an automated message if everything is working fine.

9. a. Log into your ATA. Select the "info" tab. Does "Last called Number" actually show the full phone number that you dialed?
b. Does the phone number you're trying to dial appear in Freephoneline's call logs? After you dial, is the call listed in your call log?

[Liptonbrisk]

If you're trying to spoof or fake outbound caller-ID information, you may experience issues: https://crtc.gc.ca/eng/archive/2018/2018-484.htm.
A regular 11 digit number format is required.
http://forum.fongo.com/viewtopic.php?f= ... 063#p77062

If you're using an SPA1001 without FreePBX or Asterisk (or some PBX that you haven't mentioned), I don't see how you can spoof outbound CID anyway (except for CID name) with a SPA1001 by itself.

[sic698]

I've had this service for years outbound calls stopped working as of yesterday and starting going to fast busy since yesterday, also im in another province from you.

[Liptonbrisk]

I've had this service for years

I've been using it for around 10 years. During that time, I've helped hundreds of users. Corrupted NAT associations can develop between routers and other devices on LANs, such as ATAs, without users doing anything.
That's one reason why the "Workaround for Stale NAT Mappings" section was written for your router: https://www.maxo.com.au/support/trouble ... ik-routers.

also im in another province from you.

It's free to call 519-432-1809 from any FPL account.

Again, I would go through the steps I listed.

Freephoneline customer support staff is not obligated to read posts on these forums. Technical support for problems on your end isn't free. That's why it's best to ensure the problem is not on your end first before submitting an online ticket that may be closed or ignored by default; I would confirm the problem doesn't exist with the desktop app first. I'm not saying the problem can't be a carrier issue or on FPL's end. I am saying it's best to rule out whether the issue is on your side.

[Liptonbrisk]

Another thing to check is to see whether you've accidentally enabled Caller ID block on your ATA.

[sic698]

The Status page on fongo displays

Voice Services
Incoming, outgoing, voicemail calling for Fongo Mobile/World/Works, Home Phone & FPL platforms
Degraded Performance

I don't want to troubleshoot anything if my service was working fine up until yesterday, Why should I start playing with settings when everything was fine up until yesterday

I appreciate your help though

[Liptonbrisk]

If you're trying to spoof or fake outbound caller-ID information, you make experience issues: https://crtc.gc.ca/eng/archive/2018/2018-484.htm.
A regular 11 digit number format is required.
http://forum.fongo.com/viewtopic.php?f= ... 063#p77062

If you're using an SPA1001 without FreePBX or Asterisk (or some PBX that you haven't mentioned), I don't see how you can spoof outbound CID anyway (except for CID name) with a SPA1001 by itself.

Another thing to check is to see whether you've accidentally enabled Caller ID block on your ATA.

[sic698]

Another thing to check is to see whether you've accidentally enabled Caller ID block on your ATA.

OK this seemed to have worked under line 1 --> Supplementary Service Subscription I had Block CID Serv: set to yes I put it to no and now I am able to make outgoing calls.

Thanks so much for your help

I appreciate all the effort you put into it

[Liptonbrisk]

I don't want to troubleshoot anything if my service was working fine up until yesterday, Why should I start playing with settings when everything was fine up until yesterday

I appreciate what you're saying.

Most of the things I've suggested are just part of proper setup. With the exception of changing SIP ALG settings (I'm not positive whether SIP Helper in Mikrotik routers is as buggy and horrible as SIP ALG is in other consumer routers), everything I told you should be done no matter what. That's proper setup, especially


a)Handle VIA received-->yes
b)Handle VIA rport-->yes
c)Substitute VIA Addr-->yes

That's not the problem for your current issue (for example). But that's what those ATA settings should be set to regardless.

Also, Navigate to SIP-->SIP Timer Values (sec)-->Reg Retry Intvl should be 120 seconds
(This is just a failed registration retry timer that won't help with your problem, but that's what the setting should be for Freephoneline. It wasn't included in older setup guides).
That comes from Freephoneline's own guidelines: https://support.freephoneline.ca/hc/en- ... redentials.

I appreciate your help though

I strongly suspect your issue is related to accidentally dialing *67 or a star code that blocks outbound caller ID. I'm trying to find the star code for your ATA.
Set "Block CID Serv" in your ATA to No.

[Liptonbrisk]

Another thing to check is to see whether you've accidentally enabled Caller ID block on your ATA.

OK this seemed to have worked under line 1 --> Supplementary Service Subscription I had Block CID Serv: set to yes I put it to no and now I am able to make outgoing calls.

Thanks so much for your help

I appreciate all the effort you put into it

Lol, okay, I'm glad you found the setting.

If it just suddenly happened, you or someone in your household must have accidentally dialed the star code (maybe it's *67) to enable block outgoing caller ID.