[Resolved] person trying to reach is unavailable (HT-801)

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[Resolved] person trying to reach is unavailable (HT-801)

Postby frankrizqo » 02/07/2021

Hi there,

I have been dealing with this problem for almost a year. Phone becomes unreachable, you hear a dial tone and can make calls, but cannot reach the phone. No matter where you call it from, always "the person you are trying to reach is unavailable".
To solve the problem I have to turn adapter off and then on every 1 - 2 hours or else no one can reach the phone. I am pretty sure it is an adapter problem bcz I used my sis's adapter at home and it worked fine with no interruption.
My adapter is Grandstream HT801, has been with me for 2 years so not really old. We have Rogers Ignite internet 500 mbps at home which usually works very well otherwise.

Any tip would be very helpful
I hope there is a genious out there who can solve this problem!
frankrizqo
Just Passing Thru
 
Posts: 6
Joined: 02/07/2021
SIP Device Name: Grandstream HT801
Computer OS: win 10
Router: Rogers Ignite modem (black)

Re: the person you are trying to reach is unavailable

Postby Liptonbrisk » 02/07/2021

The latest firmware version for your HT-801 ATA is 1.0.23.5 at the moment: http://firmware.grandstream.com/Release ... 0.23.5.zip
Login to your ATA and check to ensure that you are using the latest firmware version available.
http://www.grandstream.com/support/firmware.
I will not be held responsible for failed firmware updates.




Follow the steps below slowly, step by step, and please provide information when requested:


1) What brand and model modem are you using?

2) What brand and model router are you using?

a) If you are using your own router in addition to whatever modem/router combo or gateway that Rogers issued you, ensure that the Rogers device is in bridge mode.
https://www.rogers.com/customer/support ... ridgemodem

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. In particular, refer to point D in the post that follows this one, below.

3. Disable SIP ALG in whatever device Rogers gave you if you're not using your own router (or have not enabled bridge mode in Rogers' gateway).

a) For Hitron CGN3ACSMR and CODA-4582 series gateway modem/router combos from Rogers (and possibly other ISPs)
Open your web browser, and login at 192.168.0.1. Default username is cusadmin.
Select the “Basic” tab and disable “SIP ALG.” Click the “save changes” button.

b) If you have an Arris broadband gateway,
open your web browser, and login at 192.168. 0.1
Navigate to Advanced-->Options.
Uncheck the SIP box.
Click "Apply".

When new firmware updates are pushed to Hitron devices from Rogers, settings may be reset. Check to ensure SIP ALG remains disabled, periodically, or especially when you have problems.


4. Use this PDF guide to ensure your ATA is configured properly: http://forum.fongo.com/viewtopic.php?f= ... 839#p74000.
I appreciate that you have a different model ATA, but the settings should be similar.

Check all settings, especially


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


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

iii) Random RTP Port: Yes

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

v) Register Expiration is 60 minutes

vi) SIP Registration Failure Retry Wait Time: 120 seconds

vii) Enable SIP Options Keep Alive: Yes

viii) SIP OPTIONS Keep Alive Interval: 20

ix) 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 your devices now (or power cycle them in that order).


5. Ensure, after logging in at https://www.freephoneline.ca/showSipSettings that

i) SIP Status shows "connected", and
ii) 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.


6. Login at https://www.freephoneline.ca/voicemailSettings. Ensure "Rings Before Voicemail" isn't set to 1. Then click "update".

7. You may have accidentally enabled Do Not Disturb (DND). I believe with Grandstream ATAs, you dial *79 to disable DND.
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: 1781
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:5921EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: the person you are trying to reach is unavailable

Postby Liptonbrisk » 02/07/2021

(Generic info)

Typically, for VoIP SIP services, especially for Freephoneline/Fongo, you want

A) a router that does not have a full cone NAT,

Visit https://dh2i.com/kbs/kbs-2961448-unders ... -punching/.
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) < 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, 15, 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 15 for UDP Unreplied Timeout and 115 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: 1781
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:5921EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: the person you are trying to reach is unavailable (HT-80

Postby frankrizqo » 02/11/2021

Thank you soo much for the input and the detailed steps.
I tackled this systematically, upgraded the firmware successfully and then experimented with the steps one by one to see which change, specifically, will solve the problem.
This is what solved it: Enable SIP Options Keep Alive: Yes ... mind you, I did not have an option for yes!
this was based on your point (4/vii)

What I am saying is that withouth changing any of my settings from before, changing this option from "NO" which is the default to "NOTIFY" solved the problem. I have attached a picture as with the new upgrade I don't have a Yes, No option.
I chose "NOTIFY". I have not chosen "OPTIONS" although I feel like that might work too as long as it is not the default "NO".
I feel like for some reason, the defualt setting is messed up as it makes the system lose registration after a while
Attachments
voip.png
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frankrizqo
Just Passing Thru
 
Posts: 6
Joined: 02/07/2021
SIP Device Name: Grandstream HT801
Computer OS: win 10
Router: Rogers Ignite modem (black)

Re: the person you are trying to reach is unavailable (HT-80

Postby Liptonbrisk » 02/11/2021

frankrizqo wrote:Thank you soo much for the input and the detailed steps.
I tackled this systematically, upgraded the firmware successfully and then experimented with the steps one by one to see which change, specifically, will solve the problem.
This is what solved it: Enable SIP Options Keep Alive: Yes ... mind you, I did not have an option for yes!
this was based on your point



Since my FPL calls logs indicate the server accepts either OPTIONS or NOTIFY, I'm skeptical changing to NOTIFY was the real fix, unless you had No set previously. In other words, it doesn't make a difference which option you choose in this case so long as you don't select "No."

(edited from a call log)
Syslogd is listening to port# :

To: <sip:1xxxxxxxxxx@voip4.freephoneline.ca>

Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE

Supported: replaces

Event: keepalive




I feel like for some reason, the defualt setting is messed up as it makes the system lose registration after a while


Anyway, you should change 30 in your pic to 20:

Keep Alive Interval: 20 seconds*
https://support.freephoneline.ca/hc/en- ... redentials


I'm glad your issue is resolved.
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: 1781
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:5921EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin


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