Freephoneline - service down?

[carlyle705]

Since they moved the calling platform, I'm not able to call in/out anymore.

I have been with Freephoneline(Purchased SIP credential) since it was launched and I have 4 lines connected to my FreePBX server. Recently I'm experiencing huge issues with freephoneline as I cannot make and receive calls at all. Later I discovered that I can only call Fongo company number 226-336-1434 but no one can pickup my calls in billing department which I can speak to a real person. Whatever number I call, I got this message: "The number you have dialed is not in service, please check the number and try again." Whenever I call my freephoneline numbers with my cellphone, it directly goes to voice mail. I have tried to use voip.freephoneline.ca and voip2.freephoneline.ca but nothing changed.

Please tell us if the service has been closed down? I can still login my freephoneline web account with no problems.

Thanks

[carlyle705]

update:
I can call fongo mobile numbers no problem, any other numbers I can't call even toll-free numbers, same error message.

[carlyle705]

Update:

When I login to all my freephoneline web account and look at the SIP credential settings, all account shows "connected" status.

[carlyle705]

Update:

Now I have incoming calls back to work.

So far I didn't change any settings on my PBX.

[Liptonbrisk]

Always visit https://status.fongo.com/ first. If you see "degraded performance", troubleshooting is likely pointless.

The degraded performance is due to server migration: http://forum.fongo.com/viewtopic.php?f= ... 702#p76878. Anyone still registering using old server IP addresses is going to have problems.
Until the "degraded performance" status disappears, I would expect quirks.

[carlyle705]

Thanks for your reply.

I never use IP address for SIP servers, I use server names instead, voip.freephoneline.ca and voip2.freephoneline.ca are the ones I have used, but none of them worked. I'm expecting that when you change the server IPs you should change your DNS records for those known server names, but if you didn't change the DNS records, please provide your new SIP IP address.

Thanks

[Liptonbrisk]

Thanks for your reply.
I'm expecting that when you change the server IPs you should change your DNS records for those known server names

The volunteer community moderators here, including myself, don't work for Freephoneline, Fongo, nor their parent company, Fibernetics.

I'm sure DNS records have been updated, but getting the update populated everywhere can take awhile in some cases. I remember when I switched my own personal domain to a different host, getting servers worldwide to update fully took almost 24 hours.

Anyway, I think you hit the nail on the head. I suspect some services are using different DNS servers that may not have updated records yet.
I'm not having any issues, other than VMI (voicemail notifications) not reaching me yet, but it stands to reason that others might have problems for a day or two.

[carlyle705]

Thanks for your work Sorry I thought you work for Fongo.

The DNS record for voip.freephoneline.ca and voip2.freephoneline.ca seems correct as I can see the "connected" status when I login my freephoneline web account.

The problem now is that I can't call any other numbers except Fongo company number 2263361434. I even called the billing department today and they guy told me the migration is not finished yet. He asked me to wait for a few days and see. Incoming calls and voice mail are working properly for me.

[Liptonbrisk]

The problem now is that I can't call any other numbers except Fongo company number 2263361434..

What happens if you try using voip4.freephoneline.ca:6060 ?

I know you're using Freepbx, but what brand and model router are you using at home? And are you also using an ATA or IP Phone? If so, what's the model?

Anyway, it might be better to wait until the server migration is completed fully before troubleshooting.

[carlyle705]

I haven't try voip4.freephoneline.ca:6060 yet as I'm not Rogers internet subscriber. I'm using Shaw cable.

My router is Ubiquiti EdgeRouter Lite with the latest firmware.

I'm using freePBX 13 with 9 IP phones:

2 Linksys SPA962
1 Linksys SPA941
1 Cisco 7975 (SIP firmware)
2 Cisco 7961 (SIP firmware)
2 Avaya 321 (SIP firmware)
1 OBI-1022 IP phone.

They all stopped working since they started migrating to new calling system.

At this moment, incoming call seems working, but no outgoing call except fongo numbers.

[carlyle705]

I just tried voip4.freephoneline.ca:6060, didn't work. Switched back to voip.freephoneline.ca

[carlyle705]

I wish they can fix the problem as soon as possible as I have a feeling that they will discontinue the FPL service.

[Liptonbrisk]

I haven't try voip4.freephoneline.ca:6060 yet as I'm not Rogers internet subscriber. I'm using Shaw cable.

voip4.freephoneline.ca:6060 exists for potential buggy SIP ALG issues. The server has nothing to do with Rogers specifically (it doesn't matter what ISP you're with), other than their Hitron gateways contain buggy SIP ALG features in them. SIP ALG features in routers monitor traffic on UDP port 5060. voip.freephoneline.ca and voip2.freephoneline.ca uses UDP 5060. Buggy SIP ALG features in gateways, router/modem combos, hubs, and in users routers can cause many problems: The reason voip4.freephoneline.ca:6060 exists is to circumvent buggy SIP ALG features in routers and gateways.

---

Typically, for VoIP SIP services, especially for freephoneline, you want

A) 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.

B) 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.

C) 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 D) 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) < SIP OPTIONS Keep Alive Interval (in your ATA; for Obihai ATAs this is X_KeepAliveExpires) < UDP Assured Timeout (in your router) < SIP Registration Failure Retry Wait Time:(or RegisterRetryInterval in Obihai ATAs)

“<“ means less than.

When a modem leases a new IP address, a problem can arise where prior associations using the old IP address are maintained in the router. When the ATA attempts to communicate using the old IP address, the response is unreplied, and then if the UDP Unreplied timeout is greater than the Keep Alive Interval (and UDP Unreplied timeout is often set to 30 by default in consumer routers) a problem arises where the corrupted connection persists. If UDP Unreplied timeout is, for example, 17, and the NAT 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 ... 2115672/2/

The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 17 for UDP Unreplied Timeout and 117 for UDP Assured Timeout.



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.

[Liptonbrisk]

My router is Ubiquiti EdgeRouter Lite with the latest firmware.

1) Disable SIP ALG
https://docs.fusionpbx.com/en/latest/fi ... le-sip-alg

"To disable SIP ALG:

Either click on the CLI button from the Ubiquiti Edgerouter GUI or via you favorite SSH client to the Edgerouter.
Then type: configure
Then type: set system conntrack modules sip disable
Then type: commit
Then type: save
Then type: exit
"

Reboot router. Wait for it to be fully up and running first. Then reboot ATAs and IP Phones. That's always the proper device reboot order.


2) With respect to UDP timeouts (point D above), for your router the commands are

set system conntrack timeout udp stream 117
set system conntrack timeout udp other 17

stream means assured UDP timeout
other is the same as unreplied UDP timeout

UDP timeouts are not related to your current issue, however. I'm just posting instructions on how to change them if you're interested.


QoS (point C above) has nothing to do with your current problem.

At this moment, incoming call seems working, but no outgoing call except fongo numbers.

I'm able to call all phone numbers I normally call using any of my FPL numbers.

Your problem might be a FreePBX issue or a new FPL issue specific to using FreePBX. I'm not using FreePBX and don't want to set it up to troubleshoot.

[Liptonbrisk]

2 Linksys SPA962
1 Linksys SPA941
1 Cisco 7975 (SIP firmware)
2 Cisco 7961 (SIP firmware)

I'm not familiar with these devices specifically, but SPA ATAs need the following settings to work properly for two-way audio with FPL. But you're connecting using FreePBX first, so I'm not sure if there's anything you can do.

Normally for SPA ATAs connecting directly to FPL (not FreePBX), under Voice-->SIP-->NAT Support Parameters

make these changes:
a)Handle VIA received-->yes
b)Handle VIA rport-->yes
c)Substitute VIA Addr-->yes


This helps to ensure RTP audio packets are being sent to your public WAN IP address as opposed to somewhere in space (LAN IP address). Not enabling these settings can cause 1-way audio problems when using Freephoneline.

I would expect similar settings somewhere in those devices.

2 Avaya 321 (SIP firmware)

I've never heard or used it. And I don't want to look up manuals at this point.

OBI-1022 IP phone.

For OBI ATAs connecting directly to FPL,

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.


All of those devices should be using different local SIP ports.

[Liptonbrisk]

I have a feeling that they will discontinue the FPL service.

Doubtful. There would be thousands of CCTS complaints, especially if Fongo Home Phone and Fongo Mobile were still around. Every participating service provider pays a fee for each customer complaint accepted by the CCTS.
https://www.ccts-cprst.ca/


Fibernetics, which is the largest privately held CLEC in Canada, is the parent company of Fongo and Freephoneline and is, in turn, also the phone number provider for those two services.
FPL/Fongo generates revenue from incoming phone calls or termination fees to its network in addition to the fees paid by its customers. The more phone calls made to their network, the more money they make.
Given Fibernetics is a CLEC, Freephoneline doesn't need to spend money renting numbers, unlike many other VoIP service providers, and with the exception of server maintenance or needing to expand, the infrastructure should already exist. Costs are kept low.

[carlyle705]

I have a feeling that they will discontinue the FPL service.

Doubtful. There would be thousands of CCTS complaints, especially if Fongo Home Phone and Fongo Mobile were still around. Every participating service provider pays a fee for each customer complaint accepted by the CCTS.
https://www.ccts-cprst.ca/


Fibernetics, which is the largest privately held CLEC in Canada, is the parent company of Fongo and Freephoneline and is, in turn, also the phone number provider for those two services.
FPL/Fongo generates revenue from incoming phone calls or termination fees to its network in addition to the fees paid by its customers. The more phone calls made to their network, the more money they make.
Given Fibernetics is a CLEC, Freephoneline doesn't need to spend money renting numbers, unlike many other VoIP service providers, and with the exception of server maintenance or needing to expand, the infrastructure should already exist. Costs are kept low.

Thank you so much for your help!! I really appreciated this.

I already had the tweaks you mentioned above long time ago and I can tell it is not from my FreePBX. I can call Fongo mobile numbers and Fongo company number (226-336-1434), no issues. The problem is when the other party is not Fongo/FPL. When I call any numbers rather than Fongo and FPL numbers, it reports a wrong number:" The number you have dialed is not in service. Please check the number and try again.", even toll-free numbers. The message is obviously from Fongo calling system. When I call in using my rogers/telus cellphones, there is a long delay and finally went to voice mail. When I call my another FPL number, it went to voice mail immediately.

I believe there is a call routing issue in their new calling system. I have been using my FreePBX for at least 5 years now and had no issue until a few days ago they started to migrate to a new calling system.

Currently the system status still shows "Degraded Performance" and not sure what's going on about their maintenance.

[carlyle705]

today still not fixed.
Update: Incoming call from non- Fongo/FPL numbers is OK. Incoming call from FPL to FPL numbers: directly goes to voice mail.
Outgoing calls to Fongo/FPL numbers are OK, to non-Fongo/FPL numbers fails.

[Liptonbrisk]

I would suggest seeing if you have the exact same problem with Freephoneline's desktop application for Windows first after disabling FPL registrations at FreePBX 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.

If the problem persists after testing the desktop application, submit a ticket and ask for someone to check your accounts: https://support.fongo.com/hc/en-us/requests/new.
If no one responds to your support ticket after 48 business hours (they don’t work on weekends), provide the ticket number in a private message to Fongo Support: http://forum.fongo.com/ucp.php?i=pm&mode=compose&u=7852.

[carlyle705]

I have tested with the FPL app and configure a FPL line on a Linksys phone directly, they all worked!

It turns to that my current FreePBX configuration doesn't work with FPL anymore

My Version: FreePBX 14.0.13.24

Here are some suspicious lines from the asterisk log file:

VERBOSE[13320][C-000000e0] app_dial.c: Called SIP/FreePhoneLine2/8002632263
VERBOSE[11660][C-000000e0] chan_sip.c: Got SIP response 484 "Address Incomplete" back from 208.65.240.44:5060
VERBOSE[13320][C-000000e0] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
VERBOSE[13320][C-000000e0] pbx.c: Executing [s@macro-dialout-trunk:34] NoOp("SIP/106-00000178", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 28") in new stack
VERBOSE[13320][C-000000e0] pbx.c: Executing [s-INVALIDNMBR@macro-dialout-trunk:1] NoOp("SIP/106-00000178", "Dial failed due to trunk reporting Address Incomplete - giving up") in new stack
WARNING[13320][C-000000e0] channel.c: Prodding channel 'SIP/106-00000178' failed

Don't know why "Dial failed due to trunk reporting Address Incomplete - giving up", seems like FPL server doesn't like the number my FreePBX sent, but why I can call all Fongo numbers no problem?

The symptom is I can call Fongo but all non-Fongo numbers will get "The number you have dialed is not in service, please check the number and try again"
Incoming calls are fine.

Any help will be appreciated!!

Their recent maintenance has broke my FreePBX!!

[carlyle705]

A successful call log is like this:
[2020-02-02 16:18:58] VERBOSE[16574][C-000000e9] pbx.c: Executing [s@func-apply-sipheaders:14] Return("SIP/FreePhoneLine2-00000189", "") in new stack
[2020-02-02 16:18:58] VERBOSE[16574][C-000000e9] app_stack.c: Spawn extension (from-trunk, 2263361434, 1) exited non-zero on 'SIP/FreePhoneLine2-00000189'
[2020-02-02 16:18:58] VERBOSE[16574][C-000000e9] app_stack.c: SIP/FreePhoneLine2-00000189 Internal Gosub(func-apply-sipheaders,s,1(6)) complete GOSUB_RETVAL=
[2020-02-02 16:18:58] VERBOSE[16574][C-000000e9] app_dial.c: Called SIP/FreePhoneLine2/12263361434
[2020-02-02 16:18:58] VERBOSE[16574][C-000000e9] app_dial.c: SIP/FreePhoneLine2-00000189 answered SIP/106-00000188

A failed call log is like this:
[2020-02-02 17:11:32] VERBOSE[23768][C-000000ed] pbx.c: Executing [s@func-apply-sipheaders:14] Return("SIP/FreePhoneLine2-00000190", "") in new stack
[2020-02-02 17:11:32] VERBOSE[23768][C-000000ed] app_stack.c: Spawn extension (from-trunk, 8002632263, 1) exited non-zero on 'SIP/FreePhoneLine2-00000190'
[2020-02-02 17:11:32] VERBOSE[23768][C-000000ed] app_stack.c: SIP/FreePhoneLine2-00000190 Internal Gosub(func-apply-sipheaders,s,1(6)) complete GOSUB_RETVAL=
[2020-02-02 17:11:32] VERBOSE[23768][C-000000ed] app_dial.c: Called SIP/FreePhoneLine2/18002632263
[2020-02-02 17:11:33] VERBOSE[11660][C-000000ed] chan_sip.c: Got SIP response 484 "Address Incomplete" back from 208.65.240.44:5060
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] pbx.c: Executing [s@macro-dialout-trunk:34] NoOp("SIP/106-0000018f", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 28") in new stack
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] pbx.c: Executing [s@macro-dialout-trunk:35] GotoIf("SIP/106-0000018f", "0?continue,1:s-CHANUNAVAIL,1") in new stack
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] pbx_builtins.c: Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/106-0000018f", "RC=28") in new stack
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/106-0000018f", "28,1") in new stack
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] pbx_builtins.c: Goto (macro-dialout-trunk,28,1)
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] pbx.c: Executing [28@macro-dialout-trunk:1] Goto("SIP/106-0000018f", "s-INVALIDNMBR,1") in new stack
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] pbx_builtins.c: Goto (macro-dialout-trunk,s-INVALIDNMBR,1)
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] pbx.c: Executing [s-INVALIDNMBR@macro-dialout-trunk:1] NoOp("SIP/106-0000018f", "Dial failed due to trunk reporting Address Incomplete - giving up") in new stack
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] pbx.c: Executing [s-INVALIDNMBR@macro-dialout-trunk:2] Progress("SIP/106-0000018f", "") in new stack
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] pbx.c: Executing [s-INVALIDNMBR@macro-dialout-trunk:3] Playback("SIP/106-0000018f", "ss-noservice,noanswer") in new stack
[2020-02-02 17:11:33] VERBOSE[23768][C-000000ed] file.c: <SIP/106-0000018f> Playing 'ss-noservice.ulaw' (language 'en')
[2020-02-02 17:11:35] VERBOSE[23768][C-000000ed] pbx.c: Executing [h@from-internal:1] Macro("SIP/106-0000018f", "hangupcall") in new stack
[2020-02-02 17:11:35] VERBOSE[23768][C-000000ed] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/106-0000018f", "1?theend") in new stack
[2020-02-02 17:11:35] VERBOSE[23768][C-000000ed] pbx_builtins.c: Goto (macro-hangupcall,s,3)

[carlyle705]

This is really strange!!
FPL server only likes Fongo numbers!!
I can only call Fongo numbers!!

[Liptonbrisk]

Freephoneline calls to Fongo numbers are typically treated as SIP URI calls, as far as I know.

The 484 error might be a dial plan issue, but I don't use freepbx and can't help you with it. Perhaps a Freepbx user here will chime in.
You can try asking at https://community.freepbx.org/. Another good place to ask is over at https://www.dslreports.com/forum/voip.

[carlyle705]

Solved!

Liptonbrisk: Thank you so much for your help!

Note:
1. FPL doesn't allow customized caller ID anymore when you make a outgoing call to non-Fongo-FPL numbers. You have to use standard 11-digit/10-digit number format.
2. In FreePBX, this can be implemented by disable Intra-Company for the outbound route and set Force Trunk CID on each trunk.
3. When you call a non-Fongo number, only the caller ID will show up on the callee
4. When you call a Fongo number, the caller's name will show up on callee

[Liptonbrisk]

First, thank you very much for reporting back! I appreciate you taking the time to let me know. Thank you!

1. FPL doesn't allow customized caller ID anymore when you make a outgoing call to non-Fongo-FPL numbers. You have to use standard 11-digit number format.

That might be related to the enforcement of this CRTC policy: https://crtc.gc.ca/eng/archive/2018/2018-484.htm. For example, I know other VoIP service providers have enforced new rules to stop outbound number spoofing as well, in particular, after Dec. 19, 2019.

Most ATAs I've dealt with can't customize outgoing number. Only the outbound CID name can be customized.
And you can't customize the number in FPL's online portal either.

I know I used to be able to spoof outbound CID using a smartphone SIP application, but I haven't tested after server migration (also, I have no need to spoof any of my FPL numbers, so this isn't something I pursued frequently).

3. When you call a non-Fongo number, only the caller ID will show up on the callee

Have you tested to a Rogers mobile number or to a Bell landline that subscribes to CID name display?
I haven't tested outbound CID name (except to Fongo Mobile) after server migration.

Again, thank you very much for taking the time to let me know!