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Grandstream HT502 Setup Guide

PostPosted: 08/11/2014
by Jake
Credits to Val, thank you for your help.

The attached PDF will help you set up your Grandstream HT502 device from scratch. It is a known working configuration so if you are having troubles with your Grandstream HT502 reset back to defaults and follow this guide.

Edit by LiptonBrisk: Set “SIP Registration Failure Retry Wait Time” to 120 seconds.

Enabling "Allow Incoming SIP Messages from SIP Proxy Only" is a good idea for preventing SIP scanners/hackers. However, you may want to change that setting to "No because FPL to FPL and Fongo to FPL calls may not work with that setting enabled, depending on the server being used and how switches FPL uses are configured. Fongo Home Phone has this setting disabled
.

Re: Grandstream HT502 Setup Guide

PostPosted: 09/02/2016
by vostro
after HT502 upgraded firmware into 1.0.15.5(http://firmware.grandstream.com/Release ... 0.15.5.zip) automatically after a reboot, i can't get registered even i follow up this guide, it seems there are more optipons offerred in new version which i'm not sure if default settings of those new options could mess up.

Re: Grandstream HT502 Setup Guide

PostPosted: 09/02/2016
by Fongo Support
vostro wrote:after HT502 upgraded firmware into 1.0.15.5(http://firmware.grandstream.com/Release ... 0.15.5.zip) automatically after a reboot, i can't get registered even i follow up this guide, it seems there are more optipons offerred in new version which i'm not sure if default settings of those new options could mess up.

Hi,

I think you have to let Mango know.

viewtopic.php?f=8&t=16986&p=66624#p66746

Regards,

Re: Grandstream HT502 Setup Guide

PostPosted: 09/03/2016
by vostro
thanks, it looks new version firmware is buggy, pattern B is not working, instead, it's functioning like pattern C. now after factory reset, it always shows not registered but strangely i can dial out and get incoming call. it seems this vendor didn't offer any method to downgrade firmware.

Re: Grandstream HT502 Setup Guide

PostPosted: 12/20/2016
by Stoein
I bought a new ht502 to replace an old ht502.
Could somebody send me the config file for ht502
Thank you

Re: Grandstream HT502 Setup Guide

PostPosted: 12/22/2016
by Stoein
I have bought anew phone
Grandstream GXP1625
I have already registered with freephone .
So could somebody send the config file for this phone.
thankyou

Re: Grandstream HT502 Setup Guide

PostPosted: 12/22/2016
by Fongo Support
Stoein wrote:I have bought anew phone
Grandstream GXP1625
I have already registered with freephone .
So could somebody send the config file for this phone.
thankyou

Hi,

try this one:

viewtopic.php?f=15&t=14062

Very similar.

Re: Grandstream HT502 Setup Guide

PostPosted: 12/29/2016
by Stoein
Hi i have done in the setup guide
I need the string and anything else to setup the phone
thank you

Re: Grandstream HT502 Setup Guide

PostPosted: 01/04/2017
by Stoein
How long do i got wait here?
Is there anything you need from me?
thank you

Re: Grandstream HT502 Setup Guide

PostPosted: 02/12/2017
by Stoein
is freephone and fongo out of business ?
it is a long time since i heard anything
about 2 months
any buddy no what is happen
thank you

Re: Grandstream HT502 Setup Guide

PostPosted: 02/13/2017
by Liptonbrisk
Stoein wrote:is freephone and fongo out of business ?


Freephoneline and Fongo are not out of business.

Freephoneline does not provide free technical support to configure third party devices, nor are its representatives obliged to respond on forums to provide technical support for Freephoneline device configuration.

Unless you've paid $119.95 +tax for Freephoneline's VoIP unlock key, you will not be able to use Freephoneline on any device other than a computer using Freephoneline's desktop application.
http://support.freephoneline.ca/hc/en-u ... Unlock-Key

To purchase, login at https://www.freephoneline.ca/login?page ... ToPurchase.

If you've already paid for the VoIP unlock key, you can find your SIP Username and SIP Password by logging in at https://www.freephoneline.ca/showSipSettings.
Again, it is your responsibility to configure your device yourself.

If you want to contact support directly, you must submit a support ticket: https://support.fongo.com/hc/en-us/requests/new.
Alternatively, you can send a private message to Fongo Support: http://forum.fongo.com/ucp.php?i=pm&mode=compose&u=7852

Re: Grandstream HT502 Setup Guide

PostPosted: 02/13/2017
by Stoein
I have already sent a support ticket. I havent heard anything back
I have already paid unlock key
I have a disability my workers needs a l land line
if I dont hear this week I gone to bell.
I got to a have land line
Thank you

Re: Grandstream HT502 Setup Guide

PostPosted: 02/13/2017
by Liptonbrisk
Stoein wrote:I have already sent a support ticket. I havent heard anything back


They won't provide technical support for third party device configuration unless you pay $50:
http://support.freephoneline.ca/hc/en-u ... al-Support

I have already paid unlock key
I have a disability my workers needs a l land line
if I dont hear this week I gone to bell.
I got to a have land line
Thank you


Sorry, I am not familiar with that IP phone and have never used it.

Perhaps someone on the Grandstream forums can help you: http://forums.grandstream.com/forums/

Re: Grandstream HT502 Setup Guide

PostPosted: 10/01/2019
by Lydiak
Hi Everyone,

I'm hoping someone can help with a recent problem I'm having with my HT502. I've used this device for years, and starting about 6 months ago, I would experience a short period of silence while on a call. The voice would chop out for 2-5 seconds, so I often would miss a little of what the other person was saying. No issues from the other end, I tried updating the firmware to 1.0.16 but still the same issue. thanks for any help

Re: Grandstream HT502 Setup Guide

PostPosted: 10/01/2019
by Liptonbrisk
Lydiak wrote: I've used this device for years, and starting about 6 months ago, I would experience a short period of silence while on a call. The voice would chop out for 2-5 seconds


Refer to page 45 from this PDF guide: download/file.php?id=2065. I realize the PDF guide is for a different ATA, but page 45 applies to everyone, regardless of the device being used.

Re: Grandstream HT502 Setup Guide

PostPosted: 10/01/2019
by Liptonbrisk
Lydiak wrote:
I'm hoping someone can help with a recent problem I'm having with my HT502. I've used this device for years, and starting about 6 months ago, I would experience a short period of silence while on a call. The voice would chop out for 2-5 seconds, so I often would miss a little of what the other person was saying. No issues from the other end, I tried updating the firmware to 1.0.16 but still the same issue. thanks for any help



Additionally, refer to point C below.



---


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
http://www.voip-info.org/wiki/view/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, 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 ... 2115672/2/

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.