GrandStream HT-502

[Funkytown]

Your very welcome...

[linuxxtuxx]

Hello Dear Admin

I have configured my ATA box HT502 with dial plan as mentioned on my SIP settings page (with 911 dialing) but I am able to receive calls only, cannot dial any no, whenever I try after dialing the no there is a "disconnected" continuous tone only.

Please let me know what I am missing???

Thanks

Khalid

[FONGO_kris]

Do you remember the default dial plan for your Grandstream device?

If you look below the dial string it notes that it is only for Linksys / Sipura devices, once you restore this you should be able to make calls.

[linuxxtuxx]

Hi

Thanks for the reply!

I don't remember the default plan for Grandstream 502, actually I reset the device to default when my internet connection was stopped working.
Now today its suddenly start working, I can make calls but there is no voice when call is picked up.

any suggession? what to do now please let me know!!!!

Thanks

Khalid

[Funkytown]

Hi

Thanks for the reply!

I don't remember the default plan for Grandstream 502, actually I reset the device to default when my internet connection was stopped working.
Now today its suddenly start working, I can make calls but there is no voice when call is picked up.

any suggession? what to do now please let me know!!!!

Thanks

Khalid

Very Important: Anyone who doesn't have this properly put into the GrandStream 502 as there dial plan will NOT be able to recieve calls nor place calls!

Dial Plan:

Code: Select all

{911|[2-9]xxxxxxxxx|1xxxxxxxxxx|011xxxxxxxxxxxx.|*98|[6-7]x*xxxxxxxxxxx.}

[linuxxtuxx]

have put same dial plan but still no change.........

Is there any other option to change??????

Khalid

[Funkytown]

the dial plan make sure there is no space at the start nor at the end Example: {...}

please post images so we can see your settings

[linuxxtuxx]

ok, I am attaching FXS Port1 settings, the whole page comes in 4 parts which I have mentioned in the page name (1,2,3 & 4)

Also I have checked there is no space!!!

[linuxxtuxx]

Hello Dear Admin and other users

please anyone can help here??????

I am stuck in the middle as I have a working phone line but there is no voice for outgoing calls, I have pasted my device settings images as per request but still there is no reply from anyone!!!!!

I have a long distance plan but not able to use anymore because of this voice problem, please help me to find out what is I am missing or the actual fault!

my phone no is 514-4470982

Thanks....

[linuxxtuxx]

Hello to All

for God sake, please help me to adv what to do?????

I am helpless having a working phone line which can be use only to receive calls, I can dial but no voice from my side!!!!!!!!!!!!!!!!!

please help someone.......I will be really thankful to you.

Regards

Khalid

[cu2o2o2]

I really do not have a Grandstream HT and/or TP Link TD-8816, so I can not tell you where the problem is. But since nobody seems to know or offer their expertise, I will take a shot.

On your third screenshot, can you change the local RTP port from 5004 to 16348-32768? If you can, then save it and test. Also, change your dial plan to the default Grandstream HT dial plan (try to find it), as I think the symbols differ from that of Linksys (which I have). If you can not find it, use {x+}.

Let us know of progress.

[Winston]

Hey here are my 2 cents: Try removing outbound proxy and authorization ID from the grandstream, Checks the box "use random ports" in tech NAT section, click on "apply" then reboot and let's cross fingers, please let us know the results

[linuxxtuxx]

thanks guys for your kind reply!

I am getting back home by this evening, as soon as I get home, will try both options as per your adv and will post the update!

take care

Khalid

[linuxxtuxx]

hello cu2 & Winston

I have tried both of you guys but end up with same result, no change.......

Winston,

removing outbound proxy means to delete "voip.freephoneline.ca" from the third line??

The other option "use random ports" I have found but cannot find "authorization ID"

Do you mean SIP User ID or Authenticate ID???

I have tried removing both one by one but there was no change!!!

[Winston]

Yes, it is strange that they are not working, I could suggest you 2 more options:

1-make a factory reset from the ATA adapter

2-Update the firmware version.

Please let me know how is the results and I will shake my head for more ideas.

[laurent]

Here's a mini-install guide, or rather, the steps I just took to install a brand new Grandstream HT502 ATA with my FreePhoneLine account.

The HT502, when left unchecked, wants to become your network router. It could replace our typical Linksys routers and whatnot, but honestly, I want my devices to do the thing they're built for only, and have each do its own job. Therefore, I will not use mt 502 as a router or NAT, which changes a few things from how they explain it in the manual.

1. I connected the WAN port to my network, like any other device in my home. I will not use the LAN port.
2. I plugged in a phone in port #1, dialed * * *, and then followed the guide to put the unit in DHCP so that my router controls the IP allocation, and to get the MAC address.

Nota: Since it's not really a switch with 2 ports, the 502 has actually two network cards in it, and therefore, two MAC addresses. The LAN port has the MAC address that is printed on the label, and is the same as you get form the voice menus. The WAN port has the MAC address of the LAN one, +1. If it's a letter, then it goes up one (A -> B, etc). Since they're given in pairs, LAN will have an odd ending one (or A, C, E) and WAN will have even number (or B, D, F).

3. After turning DHCP on, I logged into my Linksys (with Tomato firmware) and setup a DHCP static IP for this device. Go in Basic, then Static DHCP. Use the MAC address from the +1 strategy explained in the nota above because we want the MAC of the WAN port.

tomato-1.png

4. While I'm in the Tomato firmware, let's also forward ALL ports that may be used by the VoIP adapter. Go in Port Forwarding, then Basic.

tomato-2.png

5. There's Quality of Service we could setup here to guarantee that the VoIP application has upmost priority for traffic on my internet line, but we'll do that in another post.

6. Reboot the HT502 adapter, and open a browser to the configuration page: http://192.168.0.150 (the IP I gave it). Default password is "admin"

7. Go in Basic Seetings. Only thing to change here is to set Device Mode to Bridge. Why? This will disable all NAT functions of this router, and will prevent it from trying to reroute traffic left and right.

8. Go in Advanced Settings and write a new admin password. This device controls your phone and, as such an important part of your house/life, shouldn't be left with default "please hack me" passwords.

9. Go in FXS port 1, and set it up just like this:

ht502-FPL.png

Well, that's it! I made calls, used DTMF to check my voicemail and confirm the tones work. I received a call as well (from my cellphone). Everything works out fine.

Note that, above, I picked codec G729 which is the indicated choice of FPL. Also the Dial Plan (call string) cannot be read completely in the screenshot, but it's what's been said several times in this thread, which is:

Code: Select all

{911|[2-9]xxxxxxxxx|1xxxxxxxxxx|011xxxxxxxxxxxx.|*98|[6-7]x*xxxxxxxxxxx.}

Hope this helps someone!

I also posted this guide to my blog, and I've linked back to this forum from it as well, to help as much people out as possible.

[flomar]

Today I paid $50 for configuration settings, and I was not able to make it working, I have on line1 vbuzzer and on line2 freephoneline.
I contacted customer or technical support also i wrote to freephoneline but their phone help didn't work, and now they sent me to Rogers or to device manufacturer or to reset factory default, any ideas, or can smb help.They say about port 5060 might be bloked by Rogers but that port is used by vbuzzer in their line1 setup you can check it here-http://forum.vbuzzer.com/viewtopic.php? ... 52bd63e6a8 -I read that i have to use 5062 I used no luck, so at this moment I am stuck I don't know whether or not a reset will be helpfull or I will mess my vbuzzer settings too, right now vbuzzer is working despite that the device is shown not registered.Unfortunately people who advised that it can work with two provider seemed that they didn't know what they were talking about they were speaking only based on their assumption only, not on a fgact that they tested or alike??????

[tbrummell]

SIP is not blocked by Rogers, at least not on my node.

[flomar]

I have a feeling that vbuzzer were right when they said that line2 to is worthless but i didn't take their advice,i will play for a while with device if not luck I reconsider freephoneline deal and I might ask for refund, because they assured me that the device will work with two providers.Now they blame Rogers, manufacturer or alike.

[laurent]

Flomar,

You posted your question on my blog as well, but you added one interesting piece of information: Attempting an outbound call gives you immediate busy tone. In the HT502, if the account was invalid, the voice prompts would tell you. Fast busy signal can only mean that your dial string is incorrect. Make sure you copy paste EXACTLY what I detailed in this thread:


http://forum.freephoneline.ca/viewtopic ... 579&p=8523

Make sure you copy paste the WHOLE thing, with the accolades around it { ... }

[laurent]

Oh and about ports, yes you need to set one of your SIP accounts to port 5060, and the other to 5062. Doesn't really matter which. Make sure that ports 5060, 5061, 5062 and 5063 are ALL forwarded to your device in your router. Also 130000, 130001, and 6060 and 6061 too. ALL THESE PORTS.

[flomar]

Oh and about ports, yes you need to set one of your SIP accounts to port 5060, and the other to 5062. Doesn't really matter which. Make sure that ports 5060, 5061, 5062 and 5063 are ALL forwarded to your device in your router. Also 130000, 130001, and 6060 and 6061 too. ALL THESE PORTS.

You said " Make sure that ports 5060, 5061, 5062 and 5063 are ALL forwarded to your device in your router. Also 130000, 130001, and 6060 and 6061 too. ALL THESE PORTS"
I really don't know what are this mean, i have a belkin router fsd72530-4 but I don't know what you mean.on line1 is 5060 and 5004 and on line2 is 5062 and 5012.
I already tried your setting, buts the same issue, have dial tone but when I dial after 30 sec got busy tone and the software from pc is closing up.When I am calling the number after a while got a busy tone too.
When I login again is shown fxsport2 not registered, any ideeas?
I noticed that in advance settings under Firmware Server Path and Config. Server Prefix is hwpro. vbuzzer.com/gs so vbuzzer made some settings for it too.

[laurent]

pc software? Are you using FreePhoneLine's PC based softphone at the same time? If so, turn that off. you can't have both at the same time, they will fight for your line.

Port forwards are a setting in your internet router that listens to specific port on the internet side, and if a request/message comes to that, it forwards it to one specific device in your network. This is how the outside world can reach your phone adapter (ATA). Check your router's configuration and documentation for port forwarding, and setup a port forward rule for the above mentioned ports. You can't receive calls without that.

[flomar]

I tried with the pc software turned off, same issue, I called Rogers and the same they told me to speak with belkin to find ourt how can I open ports and port forwarding, right now I will login in router to see whether or not i am smart enough to do it.Thanks anyway, I will keep you posted with my progress, if any.

[Funkytown]

flomar,laurent and to the rest here... This is what I have done... this method works flawlessly

Im using FPL config settings and what you see in these pictures is what you need to get your Voip phone working 100% with FPL http://www.voipconfigurations.com/Threa ... figuration

Have Fun

Thanks!