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

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Re: fastbusy outgoing calls only FPL

Postby Liptonbrisk » 02/05/2020

Liptonbrisk wrote:. I'm trying to find the star code for your ATA.


I found it.
According to https://www.voipmechanic.com/linksys-si ... -codes.htm, you might have accidentally dialed *67.

*68 deactivates it.
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Re: fastbusy outgoing calls only FPL

Postby sic698 » 02/05/2020

Image
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Re: fastbusy outgoing calls only FPL

Postby Liptonbrisk » 02/05/2020

sic698 wrote:Image


Yep
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Re: fastbusy outgoing calls only FPL

Postby sic698 » 02/05/2020

So I have renabled block cid service and dialed *67 and a phone number and I get fast busy if i deactivate it through *68 than dial a phone number the call goes through no fast busy.

strange
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Re: fastbusy outgoing calls only FPL

Postby Liptonbrisk » 02/05/2020

sic698 wrote:So I have renabled block cid service and dialed *67 and a phone number and I get fast busy if i deactivate it through *68 than dial a phone number the call goes through no fast busy.

strange



As I mentioned back at viewtopic.php?f=8&t=19749#p77064, I think the issue might be related to a CRTC policy that doesn't allow service providers to spoof or fake outbound Caller ID numbers with any abnormal format. Freephoneline just had a recent server migration and probably also updated rule changes on their end to coform to the CRTC policy that became enforcable on December 19, 2019. But I'm uncertain exactly what SPA ATAs are doing to SIP headers when Block CID Serv is enabled that would cause this situation to occur.
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Re: fastbusy outgoing calls only FPL

Postby sic698 » 02/05/2020

I see so there is no solution for *67 ? I cant use this function ? anymore ? I can't block my number when I call out ?
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Re: fastbusy outgoing calls only FPL

Postby Liptonbrisk » 02/05/2020

sic698 wrote:I see so there is no solution for *67 ? I cant use this function ? anymore ? I can't block my number when I call out ?


Freephoneline changed it's policy years ago. They stopped allowing *67 to work at the provider level (see #1) to prevent abuse. I think it's allowed for Fongo users though.

But I don't think you're trying to block anything at the provider level. I think you're trying to do #2 below.

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. My FPL account didn't work 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 receiving carrier that, in turn, has to recognize and agree to not show CID. *67 (outgoing CID block). Calls to Telus mobile numbers worked when *67 is processed by Obihai ATAs immediately after server migration. No CID info appeared on the receiving end. I'm not sure if *67 still works (I'm too busy to test this). But I know it did work for Obihai ATAs immediately after server migration because I tested by dialing Telus Mobility numbers.

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 CID block flag and complies. Fongo Mobile doesn't. For whatever reason *67 seems to work with Obihai ATAs, at least when calling non-VoIP numbers.

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.
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Re: fastbusy outgoing calls only FPL

Postby sic698 » 02/05/2020

Image

this is my current dial plan I will have to re-read you're last pot a few times in order to process and understand what this server migration means for me and my ATA Linksys spa 1001.

Also with callblock ID service disabled when I dial *67 I get an operator messge stating "this call is not covered by your tariff plan"
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Re: fastbusy outgoing calls only FPL

Postby sic698 » 02/05/2020

once again thanks so much for you're help Im sorry if I appeared rude on the troubleshooting process but at least we found a temporary solution
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Re: fastbusy outgoing calls only FPL

Postby Liptonbrisk » 02/05/2020

sic698 wrote:Image

this is my current dial plan I will have to re-read you're last pot a few times in order to process and understand what this server migration means for me and my ATA Linksys spa 1001.



Bottom line is that *67 isn't going to work for you.

Do you see *98 in your dialplan? That's for voicemail. When you dial *98, that code is being processed by the provider, FPL, in this case, and not by your ATA.
FPL's service knows to process *98 as voicemail. *67 does not work, at the moment, by default for the vast majority of FPL accounts when trying to have *67 processed by FPL instead of the ATA.


Anyway, you'd have to add *67 (if you wanted to see if it works for you at the provider level):

(911|[2-9]xxxxxxxxx|1xxxxxxxxxx|011xxxxxxxxxxxx.|*98|*67[2-9]xxxxxxxxx) for example

[6-7]x*xxxxxxxxxxx doesn't logically match anything that makes sense and should just be removed in my opinion.

Provided your ATA does not process *67 itself first, *67 followed by 519-123-4567 should be dialed outwards towards FPL. Because FPL does work with *67, by default, your outgoing CID information will still appear.

And we've already established that having CID Block Serv enabled in your ATA doesn't work, so #2 above won't work either.
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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Re: fastbusy outgoing calls only FPL

Postby sic698 » 02/05/2020

Image

Thats the official dialplan listed on freephonelines website when I log onto my account.

But you right about point #2 even with the dial plan changed you're dial plan setting it makes no difference caller id block service no longer works.

It's confirmed that the issue was the new CRTC policy and the server migration issue.
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Re: fastbusy outgoing calls only FPL

Postby Liptonbrisk » 02/05/2020

sic698 wrote:once again thanks so much for you're help Im sorry if I appeared rude on the trouble shooting process



You're welcome! I'm happy to help.

I'm not offended. Most people think that if they didn't do anything, a new problem can't occur on their end.
But they don't understand that corrupted NAT associations can develop in routers without the user doing anything.
NAT issues can cause all sorts of problems.

Also,


----
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 or NAT Keep Alive Intvl (in your ATA; for Obihai ATAs this is X_KeepAliveExpires; NAT Keep Alive Intvl is supposed to be 20 seconds according to FPL guidelines) < UDP Assured Timeout (in your router) < SIP Registration Failure Retry Wait Time (or Reg Retry Intvl in SPA ATAs . . . this is supposed to be set to 120 seconds according to FPL guidelines)

“<“ 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.

Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers may be difficult, if not impossible. Asuswrt-Merlin, third party firmware for Asus routers, does offer easy access to these two settings, which are found under Tools-->Other settings. In part, for this reason, I tend to use Asus routers. However, 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.

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.
---

You have a Mikrotik router. You can adjust UDP timeout settings: https://forums.redflagdeals.com/recomme ... #p28056619


So NAT corruption in routers can occur without a user doing anything. Improper UDP timeouts can also be another issue that can cause problems without a user doing anything.
If you never experience problems, then don't change anything, but what I've posted here is something to keep in mind in the event a problem does arise in the future.

Also, issues involving choppy audio can be on a user's end due to a lack of implementing QoS for an ATA or IP Phone (giving traffic priority to the VoIP device instead of allowing other devices on a LAN to hog available bandwidth at the same time someone is trying to make a call). Choppy audio or severe jitter can also be an ISP issue.
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Re: fastbusy outgoing calls only FPL

Postby Liptonbrisk » 02/05/2020

sic698 wrote:Image

Thats the official dialplan listed on freephonelines website when I log onto my account.


Yeah, that [6-7]x*xxxxxxxxxxx has never made any sense to me.
That means match any two digits beginning with 6 or 7 (65 for example), followed by a *, and then 11 digits.
lol What is that for?????

65*15191234567

or

79*14161234567

Makes no sense. I almost venture to say that's a mistake. Maybe there's some weird dial string that matches that pattern and can be processed by FPL, but it does not match anything that I can think of.
I have omitted [6-7]x*xxxxxxxxxxx every time I setup FPL for users.
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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Re: fastbusy outgoing calls only FPL

Postby Cyber » 02/05/2020

I confirm that I cannot use anymore "Block CID Serv" to make outgoing calls to mobile phones.

Before having the fast busy tone, I had a 15 minutes call disconnection issue by using that feature.
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Re: fastbusy outgoing calls only FPL

Postby Liptonbrisk » 02/05/2020

Cyber wrote:I use that dial plan on my SPA112 :

<311:5148720311>S0|<411:18005551212>S0|<511:18883550511>S0|<811:18003613977>S0


The colons and the phone numbers following them aren't necessary for you anymore: https://www.fongo.com/government-service-numbers/.

[2-3]11 works in Quebec and Montreal, respectively
411 should work, but it costs money. I'm not testing it.
[5-6]11 works in Quebec
[8-9]11 works in Quebec
Those specific N11 codes are handled at the provider level now without the ATA needing to transpose numbers.

811 in Ontario requires <811:8667970000> for Telehealth Ontario at the moment.


Before having the fast busy tone, I had a 15 minutes call disconnection issue by using that feature.


15 minute before disconnect makes no sense at all. There's no reasonable explanation for it.
Immediate call fail or busy makes more sense.
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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Re: fastbusy outgoing calls only FPL

Postby Cyber » 02/05/2020

Liptonbrisk wrote:The colons and the phone numbers following them aren't necessary for you anymore: https://www.fongo.com/government-service-numbers/


Thanks for the information. I updated my SPA112 dial plan to :

(911S0|211S0|311S0|511S0|811S0|*xxS0|[2-9]xxxxxxxxxS0|1xxxxxxxxxxS0|011xxxxxxxxxxxx.|411!|1900x.!|1976x.!)
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SPA/PAP *67 Anonymous call 15 minute disconnect

Postby albeause » 04/08/2020

I have the same issue even with using voip4.freephoneline.ca:6060, as i didn't find the option SIP-ALG on my router (Helix- FI gateway CGM414X). Reboot has been done in sequence, no luck.
It does the same with voip.freephoneline.ca, drops after 15 mins.

So calling FPL to FPL ,, works after 15mins.
Calling from cellphone (videotron) to FPL,, works after 15mins.
Calling from FPL to cellphone (videotron), call drops after 15mins.

I hope that issue will be resolved soon as i cannot use my FPL to call on my conference calls meetings as it's disconnect.
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Re: Calls are mute after 15 minutes

Postby Liptonbrisk » 04/08/2020

albeause wrote:I have the same issue


I think a SIP re-invite is occurring at the 15 minute mark and not being acknowledged, but maybe something else is happening.


You might consider submitting a ticket and asking about the 15 minute drop due to what I'm going to write way down at the bottom of this post. In the ticket I would mention that you wonder whether due to server load, calls have been dropping after 15 minutes. In the ticket, I would mention the proxy server you're using and the time of the call that dropped. If you can repeat the problem over and over again when calling a specific phone number, provide the phone number in the ticket. I suspect, though, you'll find the problem is random.

even with using voip4.freephoneline.ca:6060, as i didn't find the option SIP-ALG on my router


voip4.freephoneline.ca:6060 should be bypassing SIP ALG if it's on (hidden setting) in your router. Some routers have SIP ALG stuck on with no way to disable it. That's what voip4.freephoneline.ca:6060 is for.

Also,

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 your SPA2102, 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




So calling FPL to FPL ,, works after 15mins.
Calling from cellphone (videotron) to FPL,, works after 15mins.
Calling from FPL to cellphone (videotron), call drops after 15mins.


Really, that seems almost random.

I hope that issue will be resolved soon as i cannot use my FPL to call on my conference calls meetings as it's disconnect.


The thing is that 3 people in this thread appear to have resolved their issue. I haven't been able to reproduce it. I'm going to test on voip4.freephoneline.ca:6060 for you shortly.

viewtopic.php?f=8&t=19765#p77199
JOHN-SBK wrote:A quick update to say I just completed a 1h36min call without any problem. Great!


viewtopic.php?f=8&t=19765&start=25#p77289
mkaye wrote:a friend sent his incoming/outgoing config
i changed a few parameters and now it is FIXED!


viewtopic.php?f=8&t=19765&start=25#p77295
Jos wrote:@mkaye it works now to pass the 15 minutes, the error I made is when I used the copy and paste command I did not include the last command
canreinvite=no

Thank you very much. :D :D :D


And then there's
viewtopic.php?f=8&t=19734#p77009 (which doesn't make any sense, really; I still feel that's coincidental and probably random)


Anyway, I haven't been able to reproduce the 15 minute dropped call issue at all. And I would normally chalk this up to an issue on the user's end (configuration issue or lack of QoS) or with the user's ISP (packet loss leading to dropped calls).

EXCEPT

Someone at Fongo Support acknowledged voip.freephoneline.ca was seeing a lot of call volume. The dropped call problem almost seems as though this problem is random. Maybe it has nothing to do with you calling your cellphone.
Maybe it has to do with server load. That would make the issue intermittent. That is, when load is high, calls are dropped. When load isn't high, calls last a long time.

I'm uncertain why I've never encountered this issue, but maybe I haven't been using a server when there was high call volume on it.

Anyway, I'll try testing on voip4.freephoneline.ca:6060 now, which may be meaningless right now if the server isn't being heavily used.
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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Re: Calls are mute after 15 minutes

Postby Liptonbrisk » 04/08/2020

Okay, I'm at the 19 minute mark of calling a regular mobility number while registered on voip4.freephoneline.ca:6060 without having the call drop.

So, if there is a call duration issue involving server load, I've never encountered it. I've certainly placed long calls (over an hour) while registered on voip2.freephoneline.ca and voip.freephoneline.ca previously.

That doesn't mean a problem involving congestion doesn't exist though. It just means I've never experienced the problem. I've tried to reproduce the problem every time a user has mentioned it.
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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Re: Calls are mute after 15 minutes

Postby albeause » 04/09/2020

Hi !
On my side i think i found the issue. The problem seems to occurs when *67 is activated,, meaning when calling from FPL and not showing the CallerID on the other end.
When *67 is activated from FPL to call my cellphone, the call is dropped in 15 mins.
When *68 is activated (showing CallerID), and calling my cellphone, it goes beyong the 15 mins.
Those test i did, i was on proxy voip.freephoneline.ca
@Liptonbrisk,, if you have a chance to test it, i'm curious to see your result.
Thanks !
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Re: Calls are mute after 15 minutes

Postby Liptonbrisk » 04/09/2020

albeause wrote:Hi !
On my side i think i found the issue. The problem seems to occurs when *67 is activated,, meaning when calling from FPL and not showing the CallerID on the other end.
When *67 is activated from FPL to call my cellphone, the call is dropped in 15 mins.
When *68 is activated (showing CallerID), and calling my cellphone, it goes beyong the 15 mins.
Those test i did, i was on proxy voip.freephoneline.ca
@Liptonbrisk,, if you have a chance to test it, i'm curious to see your result.
Thanks !



Well, now you're the second person to say that *67 causes the call to drop after 15 minutes. That still doesn't make any sense to me because if a call is rejected based on *67, the call should be rejected immediately and not after a SIP re-invite at 15 minutes.

Okay, I'll test. I did test before here: viewtopic.php?f=8&t=19734#p77014. But I can't remember what proxy server I was using. So, I'll test again and make sure to use voip.freephoneline.ca.
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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Re: Calls are mute after 15 minutes

Postby Liptonbrisk » 04/09/2020

Okay, I'm over 18 minutes now calling a cellphone number using voip.freephoneline.ca registered on an Obihai ATA. CID appears as "No Caller". *67 was processed by the ATA (as opposed to having *67 sent first to FPL).

Call hasn't dropped.

On April 3rd, I had some calls that were over 40 minutes long, but I wasn't using *67.


Are you positive the 15 minute drop occurs every single time you use *67? If so, maybe Cyber is right. Maybe the problem only affects Linksys/Cisco devices when *67 is used. But if that is the case, it doesn't make any sense why the call isn't dropped immediately instead.

Anyway, I find this situation difficult to debug if I can't reproduce it.

It may be worth capturing a syslog and sending the contents of it in a ticket to Fongo:
https://www.cisco.com/c/en/us/support/d ... 08784.html
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
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Re: Calls are mute after 15 minutes

Postby albeause » 04/11/2020

Hi !

To answer your question,, yes it drops in 15 mins every single time, when the *67 is activated. I tried FPL to Bell lan line, FPL to cellphone Videotron and FPL to Bell Mobility. All three does the same result after 15mins mark,, the call dropped.

I did capture syslogs, with spa2102 set in full debug, and it does not show the issue. The SIP/2.0 200 Ok and NOTIFY sip messages are still transmitting from the ATA to the proxy even the call is dropped on the cellphone (which it could be normal to continue the registration). There is no special log when the call drop. I remark is when the disconnect occurs on the cellphone on the 15 mins, there is no hang up logs (BYE Sip, CC: ended, FM Alert Stop Rx Tx, or AUD rel Call). Normally without *67, (*68 activated) when the other end hangs up,, you get these logs and also you will get the busy tone, followed with a fast busy tone. You don't hear any tone when *67 is activated, it comes dead-air.

Maybe FPL team will have to pull CDR on the switch to find out where is the disconnect. I will file a ticket and see.

Thanks for your help !
albeause
Just Passing Thru
 
Posts: 15
Joined: 12/28/2016
SIP Device Name: spa-2102
Firmware Version: 5.2.13
ISP Name: Videotron
Computer OS: Windows 10
Router: Helix FI gateway CGM414X

Calls are mute after 15 minutes

Postby Liptonbrisk » 04/11/2020

albeause wrote:Maybe FPL team will have to pull CDR on the switch to find out where is the disconnect. I will file a ticket and see.


15 minute drop issues are almost always due to an ACK not being received after an Invite is requested. SIP ALG problems for FPL users used to cause calls to drop after 32 seconds in the past. As for why this is now happening due to using *67, after server migration, I have no clue. If a call is supposed to be dropped due to using *67, it should be dropped immediately instead.

Maybe create a .zip file containing the syslog file and password protect it. Upload it somewhere, and give Fongo Support the link to the file (along with the password) in the the ticket.
There's nothing else that regular FPL customers like us can do. I don't know that anything will be done to address a 15 minute call drop due to using *67 on PAP/SPA devices, but it doesn't hurt to ask.

Yes, the switch vendor they use would probably have to address this. Based on what you've written, it seems possible, to me, that a bug was introduced during the server migration.

Thank you
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: 2764
Joined: 04/26/2010
SIP Device Name: Obihai 202/2182, Groundwire
Firmware Version: various
ISP Name: FTTH
Computer OS: Windows 64 bit
Router: Asuswrt-Merlin & others

Re: *67 Anonymous call/CID Block Serv fast busy/15 minute dr

Postby albeause » 04/15/2020

Hi !
I submit a ticket and provided syslogs and a few days later they close the ticket without any comment. I replied that the issue was still present, The next day i got an automated email message:
"We have received your Freephoneline support request.
Please note that we do not provide free support for Freephoneline, but there is an active community that provides support to fellow Freephoneline users for all matters from the desktop app to configuring your own ATA with the VoIP unlock key. You can check it out by visiting http://fongo.help/fpl
We do have pay-per-incident support for Freephoneline, find out more by visiting http://fongo.help/fplppi
If you would like our supported home phone please visit http://www.fongo.com/home-phone or call 1-855-553-6646 x 309 to talk to a sales representative."

I won't pay 50$ to get the system fixed. I prefer to spend the money to a more recent ATA (spa2102 seems not supported) and as per the tests you did, you were unable to reproduce, it could be that my equipment is too old.

Thank you very much for your help !
albeause
Just Passing Thru
 
Posts: 15
Joined: 12/28/2016
SIP Device Name: spa-2102
Firmware Version: 5.2.13
ISP Name: Videotron
Computer OS: Windows 10
Router: Helix FI gateway CGM414X

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