Grandstream HT-801: Inbound calls go directly to voice mail

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Grandstream HT-801: Inbound calls go directly to voice mail

Postby Anderus7 » 08/30/2020

Grandstream HT-801: Inbound calls go directly to voice mail after about 180 minutes, after reboot.

I recently replaced my old Linksys spa2102 with Grandstream HT-810. I setup the new ata according to a pdf file listed in this forum for HT-701. Everything seems to work fine except after 3 hours or so, attached phone does not ring for an incoming call. Calls go directly to FPL’s voice mail.

I have tried using:
Primary SIP Server: voip4.freephoneline.ca
Local SIP Port: 6060.
It did not help. I am baffled.

Teksavvy modem is: CDA3-35.

Your help is very much appreciated.
Anderus7
One Hit Wonder
 
Posts: 1
Joined: 08/30/2020
SIP Device Name: Grandstream HT-801
Firmware Version: 1.0.19.11
ISP Name: Teksavvy
Computer OS: Windows 10
Router: TP Link AC750 C2

Re: Grandstream HT-801: Inbound calls go directly to voice m

Postby Liptonbrisk » 08/30/2020

Primary SIP Server: voip4.freephoneline.ca
Local SIP Port: 6060


Follow the steps down the list, carefully, step by step.

i) Primary sip server should be "voip4.freephoneline.ca:6060" without the quotation marks.
6060 here has nothing to do with local sip port.

ii) Choose a random number between 30000 and 60000 for the local sip port.

Local SIP port is separate from the primary sip server port.

iii)Use random sip port should be set to yes.



Ensure

iv) Random RTP Port: Yes

v) SIP REGISTER Contact Header Uses is set to WAN address

vi) Register Expiration is 60 minutes

vii) SIP Registration Failure Retry Wait Time: 120 seconds

viii) Enable SIP Options Keep Alive: Yes

ix) SIP OPTIONS Keep Alive Interval: 20


A) Disable SIP ALG in TP Link AC750 C2

Login to your router.
Navigate to Security-->Basic Security-->disable SIP ALG.
Click Save

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. Reboot devices now.


B) When the problem occurs, ensure, after logging in at https://www.freephoneline.ca/showSipSettings, that

a) SIP Status shows "connected", and
b) SIP User Agent reflects a device that own and recognize. If you don't recognize the SIP User Agent, chances are you've been hacked.

Note that 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, consequently, incoming calls will not work on it. This is especially important to consider if someone else is using your SIP credentials (username and password) that are found after logging in at https://www.freephoneline.ca/showSipSettings or if you're trying to register your FPL account with a smartphone SIP app or with another device. Registration is required for incoming calls. It is not required for outgoing calls. A more significant concern, though, is that multiple registration attempts can lead to temporary IP bans. The more devices being used can make the temporary ban happen more quickly. Note that each time you reboot or restart your ATA or SIP app, it's attempting to register with Freephoneline again. Multiple registration attempts within a short period can result in temporary IP ban. Each time you reboot your ATA it's attempting to register with FPL's proxy server.

C) Log in at https://www.freephoneline.ca/doGetCallLogs. Select the current month, and select "submit." See if the call in question is listed.

D) If the incoming call is coming from another FPL or Fongo number, try disabling "Allow incoming SIP Messages from SIP Proxy only" in the ATA to see if that resolves the issue.
FPL and Fongo to FPL calls are treated as SIP URI calls. The purpose of "Allow incoming SIP Messages from SIP Proxy only" is to help thwart hackers, but I'm uncertain whether enabling that setting currently blocks incoming calls from Fongo Mobile or Fongo Home Phone users, for example.

E) Take a look at point #4 below.

Also, for #3, it is possible for incoming calls to not ring due to the problem described with respect to lack of QoS for your ATA. This might help (I'm not familiar with your router and don't have time to read the manual on it): https://www.tp-link.com/ca/support/faq/1104/. If you need help configuring QoS for your ATA, you can try asking at https://community.tp-link.com/en/home/forum/32.

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.
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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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: Grandstream HT-801: Inbound calls go directly to voice m

Postby Liptonbrisk » 08/30/2020

(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; for Grandstream, the setting is SIP OPTIONS Keep Alive Interval) < 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.

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

Asuswrt-Merlin
Ubiquiti
Mikrotik
pfSense
Tomato
DD-WRT


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



ISPs do not issue customers routers that can do all four things I just listed. Typically it's far better to have your own router with strong QoS functions and a restricted cone NAT firewall,
disable whatever SIP ALG feature is enabled in the router, and stick whatever modem/router combo your ISP gives you into bridge mode. For Bell Hubs, visit http://forums.redflagdeals.com/please-s ... r-1993629/. For Rogers, visit https://www.rogers.com/customer/support ... ridgemodem.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
User avatar
Liptonbrisk
Technical Support
 
Posts: 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: Grandstream HT-801: Inbound calls go directly to voice m

Postby grrizzlly » 09/04/2020

Same issue here on HT-801 and HT-802, both worked fine at 2 different locations till couple of days ago. Outbound calls work fine. SIP shows as registered and incoming calls show in call log. All incoming calls go to v/mail.
grrizzlly
Just Passing Thru
 
Posts: 16
Joined: 03/28/2015
SIP Device Name: Linksys SPA2102
Firmware Version: 5.2.13(004)
ISP Name: Start.ca

Re: Grandstream HT-801: Inbound calls go directly to voice m

Postby Liptonbrisk » 09/04/2020

grrizzlly wrote:Same issue here on HT-801 and HT-802, both worked fine at 2 different locations till couple of days ago. Outbound calls work fine. SIP shows as registered and incoming calls show in call log.


Is there a disconnect reason listed in FPL's call log from the website?


All incoming calls go to v/mail.



1)Did you follow all the steps from viewtopic.php?f=8&t=19918#p77890 ? For example, are you using voip4.freephoneline.ca:6060?

2) What brand and model modem are you using?

3) What brand and model router are you using?

For those that are absolutely convinced the problem is with FPL, submit a ticket: https://support.fongo.com/hc/en-us/requests/new.
Maybe enough tickets will convince them to look at a possible server congestion/overload issue.

Until I can reproduce the problem (and I've tried), there's not much I can do.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
User avatar
Liptonbrisk
Technical Support
 
Posts: 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: Grandstream HT-801: Inbound calls go directly to voice m

Postby grrizzlly » 09/04/2020

Normal call clearing is disconnect reason from the log. Let me get in and check config.
grrizzlly
Just Passing Thru
 
Posts: 16
Joined: 03/28/2015
SIP Device Name: Linksys SPA2102
Firmware Version: 5.2.13(004)
ISP Name: Start.ca

Re: Grandstream HT-801: Inbound calls go directly to voice m

Postby grrizzlly » 09/04/2020

Primary server was voip.freephoneline.ca. Changed it to voip4.freephoneline.ca:6060 and it did the trick.
grrizzlly
Just Passing Thru
 
Posts: 16
Joined: 03/28/2015
SIP Device Name: Linksys SPA2102
Firmware Version: 5.2.13(004)
ISP Name: Start.ca

Re: Grandstream HT-801: Inbound calls go directly to voice m

Postby Liptonbrisk » 09/04/2020

grrizzlly wrote:Primary server was voip.freephoneline.ca. Changed it to voip4.freephoneline.ca:6060 and it did the trick.


Great. Thanks for let me know.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
User avatar
Liptonbrisk
Technical Support
 
Posts: 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


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