Calls randomly going to voicemail with Grandstream ATA

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Calls randomly going to voicemail with Grandstream ATA

Postby THeadphones » 05/27/2021

I ran in to an issue recently where roughly half my calls appear to be going to voicemail. I haven't pinpointed this to a specific vendor yet, but nothing has changed on my Router/Internet Provider/SIP Device other than trying different voipX.freephoneline.ca Proxy Servers.

Had been problem free for over a year. Any advice for troubleshooting this? Calling the line from my cell seems to go 100% through, calling the line from my office seems to go 100% to voice mail.
THeadphones
Just Passing Thru
 
Posts: 5
Joined: 08/06/2020
SIP Device Name: Grandstream HT801

Re: Calls randomly going to voicemail

Postby Liptonbrisk » 05/28/2021

You originally posted in the Fongo Home Phone forum, which you are not using. I moved your thread.

Due to configuration changes made with FPL's switches, it is now important that SIP ALG be disabled completely in your router(s). Alternatively, use voip4.freephoneline.ca:6060.
Refer to point B below.

This is unrelated because something has changed. However, take a good look at point D below as well. Issues can occur in users' routers, such as NAT corruption, without them doing anything. Consequently, users claiming that nothing has changed on their end doesn't mean very much. Additionally, ISPs can push firmware updates, sometimes causing settings to reset or change, in user's gateways or hubs without their knowledge, which is troublesome if SIP ALG becomes enabled.

In the next post, I've listed steps for you to follow.

--
(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 ... #p28059363.

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.
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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: Calls randomly going to voicemail

Postby Liptonbrisk » 05/28/2021

1) What brand and model modem are you using?

If you're using a modem/router combo, gateway, or hub issued by your ISP, contact your ISP to ask for assistance for disabling SIP ALG.
If you are also using your own router in addition to the one supplied by your ISP, then you should be enabling bridge mode instead; refer to step 2 below.

Hitron CGN series gateway modem/router combos (from Rogers, Shaw, or another ISP) or any modem/router combo from any ISP with SIP ALG forced on

If you don’t have your own router, and if you can’t get someone from Rogers or your ISP to disable SIP ALG for you in their modem/router combo, your ATA may need to register with voip4.freephoneline.ca:6060. The purpose of voip4.freephoneline.ca:6060 is to help circumvent faulty SIP ALG features in routers. So, if you’re experiencing one-way audio issues or calls going to voicemail as a result of SIP ALG, this is the SIP server to use. Check to ensure that you can’t disable SIP ALG yourself.


Hitron CGN3ACSMR and CODA-4582 series gateway modem/router combo 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.


Arris XB6 from Rogers

Open your web browser, and login at 192.168. 0.1
Navigate to Advanced-->Options.
Uncheck the SIP box.
Click "Apply".



2) What brand and model router are you using?
a) Make sure whatever modem/router combo, gateway, or hub your ISP gave you is in bridge mode if you are using your own router. Call/contact your ISP if you have to.

This is how you enable bridge mode with Rogers' gateway devices: https://www.rogers.com/customer/support ... ridgemodem.
Only do this if you are also using your own router in addition to Roger's gateway device; it is not just a modem; it is modem/router combo.


3) Make sure you disable SIP ALG if you're using own router. Here is an example:
https://www.obitalk.com/info/faq/sip-alg/disable-alg. If you wish for more specific help, I need the brand and model of the router you're using.

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).

4) What brand and model ATA are you using?
Your profile lists Grandstream HT-801.
I suggest following the settings listed in this guide as much as possible: viewtopic.php?f=15&t=18839
Particularly, ensure SIP REGISTER Contact Header Uses is set to WAN address.

Obviously, only use the firmware for your own specific ATA model. Grandstream ATA firmware is located at http://www.grandstream.com/support/firmware.
Check for a firmware update.

5) Use voip4.freephoneline.ca:6060 for Primary SIP server

6) Lastly, this is always proper device reboot order:

A.Turn off modem, router and ATA.

B. Turn on modem. Wait for modem to be fully up and running.

C.Turn on router.
Wait for modem to be fully up and transmitting data before turning on router.

D. Turn on ATA only after the router is fully up and running.

Reboot your devices now.

7) Test with incoming calls
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: Calls randomly going to voicemail with Grandstream ATA

Postby THeadphones » 06/08/2021

My cablemodem is a Hitron CODA-4582 with the routing disabled & am running a Asus AC-5300 w/a fair bit of tweaks to it, but I can make any requested changes. What settings out of the box should it be using?

Is SIP ALG a router setting or a modem setting, and is the Hitron capable of messing with stuff with this option on with routing disabled?

I can flash the cablemodem to whichever firmware is capable of correcting this, but as I said previously I was running it for over 6 months before this started becoming an issue.

Thanks for any additional input, will look in to the above further soon
THeadphones
Just Passing Thru
 
Posts: 5
Joined: 08/06/2020
SIP Device Name: Grandstream HT801

Re: Calls randomly going to voicemail with Grandstream ATA

Postby Liptonbrisk » 06/09/2021

THeadphones wrote:My cablemodem is a Hitron CODA-4582 with the routing disabled


Follow the steps listed here, step by step, down the list, slowly and accurately:

1. Visit https://www.rogers.com/customer/support ... e-coda4582.
Ensure Residential Gateway Function is disabled.
That's bridge mode. Use bridge mode (disable residential gateway function).

am running a Asus AC-5300 w/a fair bit of tweaks to it, but I can make any requested changes. What settings out of the box should it be using?


I would strongly consider switching to Asuswrt-Merlin, which is what I use: https://www.asuswrt-merlin.net/.
Merlin firmware downloads for your RT-AC5300 router can be located at https://sourceforge.net/projects/asuswr ... 0/Release/
Check https://www.asuswrt-merlin.net/ for the latest non-beta release version first, and then download the appropriate file.
Save your router settings, and backup jffs partition first before flashing to Merlin, just in case you want to flash back to official Asus firmware afterwards.
You can then just restore your settings.


After flashing to Merlin, reset router to factory defaults.
Then make whatever changes you want.
I will not be held responsible for failed firmware updates.
It's your choice whether you switch to Merlin, but Merlin makes sense for people who use SIP services, such as Freephoneline.


Afterwards, refer to point D from https://forum.fongo.com/viewtopic.php?f=8&t=20189&p=78967#p78930 concerning Unreplied and Assured UDP timeouts. In Merlin, you can adjust those settings by navigating to General–>Tools-->Other settings. Both settings are not available in official Asus firmware from the web browser UI.

If you are hesitant or have concerns before installing Merlin, you can voice your concerns and ask questions here: https://www.snbforums.com/forums/asuswrt-merlin.42/.

I didn't provide a numbered step here since this section is optional. However, if you ever do encounter problems with incoming calls that can't be resolved without rebooting the ATA (and/or router), you'll know what you should be doing instead of having to reboot devices.

Is SIP ALG a router setting or a modem setting


Router. However, Hitron CODA-4582 isn't just a modem. It's a modem/router combo.The router features in it (SIP ALG) cause problems.
When bridge mode is enabled (which is the same as having Residential Gateway Function disabled in the Hitron CODA-4582), the SIP ALG setting should be turned off automatically, which is a good thing, and will not be available.

2. Login to Asus router, navigate to Advanced Settings-->WAN-->SIP Passthrough (this is the SIP ALG setting in Asus routers)-->Select "Disable".
Click "Apply".

In Merlin the ALG is just "+NAT Helper". So, as long as you don't have "Enabled + NAT Helper" selected for SIP Passthrough, you should be fine.

I can flash the cablemodem to whichever firmware is capable of correcting this, but as I said previously I was running it for over 6 months before this started becoming an issue.


Well, you're having a problem now. NAT corruption can occur in users' routers without them changing anything. Refer to point D from https://forum.fongo.com/viewtopic.php?f=8&t=20189&p=78967#p78930 concerning Unreplied and Assured UDP timeouts. Rogers can also push firmware updates to your Hitron modem/router combo without notifying you, which can cause changes to be made.

Moreover, configuration with FPL's switches appears to have changed, and it's now important to get SIP ALG (SIP Passthrough) disabled for some users. That is likely related to the issue you're experiencing. See here: viewtopic.php?f=8&t=20182. Disabling SIP Passthrough (which is SIP ALG) was the solution.

3. What brand and model ATA are you using?
Your profile lists Grandstream HT-801.
I suggest following the settings listed in this guide as much as possible: viewtopic.php?f=15&t=18839
Particularly, ensure SIP REGISTER Contact Header Uses is set to WAN address.

Obviously, only use the firmware for your own specific ATA model. Grandstream ATA firmware is located at http://www.grandstream.com/support/firmware.
Check for a firmware update.

4. Use voip4.freephoneline.ca:6060 for Primary SIP server in your ATA. This helps to ensure SIP ALG is not causing a problem.
The purpose of voip4.freephoneline.ca:6060 is to circumvent SIP ALG in routers. SIP ALG monitors traffic on UDP port 5060 and mangles SIP headers.

5. This is always proper device reboot order:

A.Turn off modem, router and ATA.

B. Turn on modem. Wait for modem to be fully up and running.

C.Turn on router.
Wait for modem to be fully up and transmitting data before turning on router.

D. Turn on ATA only after the router is fully up and running.

Reboot your devices now.

6. Test with incoming calls


Lastly, I have configured an ATA for someone using a Rogers Hitron CODA-4582U (white cube) in bridge mode with an Asus router running Asuswrt-Merlin (+NAT Helper is disabled for SIP Passthrough). All incoming calls work. FPL to FPL calls work. Regular landline to FPL calls work. Mobility calls to FPL work.
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: Calls randomly going to voicemail with Grandstream ATA

Postby THeadphones » 06/29/2021

Super appreciative of the lengthy guide!

I plan on going through it this Thursday (holiday) & should have time to tinker with things. I'm currently using a OBI-110, as I don't have access to the Fongo provided HT-801. I had configured it using one of the guides on the forums. It may be worth noting that the calls going to voicemail do not show up on the OBI-110 (they do however, show up on freephoneline.ca's call log).

Will post up in a couple of days with the results!

Thanks again
THeadphones
Just Passing Thru
 
Posts: 5
Joined: 08/06/2020
SIP Device Name: Grandstream HT801

Re: Calls randomly going to voicemail with Grandstream ATA

Postby Liptonbrisk » 06/30/2021

THeadphones wrote:S

I plan on going through it this Thursday (holiday) & should have time to tinker with things. I'm currently using a OBI-110, as I don't have access to the Fongo provided HT-801.



Well, then step 3 is using this guide: viewtopic.php?f=15&t=16196#p64070. The only difference between your setup and others is that most people I know are using an Obihai OBI2xx series ATA and not the older 1xx series.

OBi2xx series ATAs have a setting called X_UsePublicAddressInVia (and that setting is enabled in the configuration guide), but I don't think that setting is available in OBi1xx series ATAs.
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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