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Grandstream GWX-410x


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Hi everyone.


I've been trying for the past 3 days to set up the Grandstream GWX-410x gateway with my installation of PBXnSIP. I have tried everything and I cannot get any incoming or outgoing calls to work. Here is what I have done so far. If anybody can help me out with this, I would really appreciate it.


The IP Address of the gateway is

The IP Address of the PBX Server is


I set up a Trunk called PSTN Gateway. I set the Outbound Proxy to the IP of the gateway. Under CO Lines I have co1 co2 co3 co4. The extension is set to 70 which is our auto-attendant. Everything else I left as default. I'm not sure what to put as the username or password. I left the username blank and tried the administrator password for the gateway and also the user password.


I set up a dial plan to use this trunk. The pattern is 9* and the replacement is empty.


At the gateway, I left almost everything to the default values. But I changed to one stage dialing. Under profile 1, I set the IP Address of the SIP Server and clicked NO to SIP Registration. Everything else is default.


Please Please Please someone help me out. I have tried just about everything and I'm not sure why this is not working. I really would appreciate your help.




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What version of the firmware are you using?


Just a few general pointers on the Grandstrream FXO that I discovered after many hours of testing and observing.

Some make sense and some don't. I guess its that way until you & I understand the unit better. Whenever you re-boot the Grandstream make sure it does re-boot by watching the LEDS. If they all don't go out and then back on again, it did not re-boot. (even if they do, it still may not be updated - at least thats what I found to be the case) I set up a Syslog Server (KIWI) on another computer and it says something like

"channel: 0 is busy, user req of reboot is postphoned for 10s.". So if you don't see the LEDS go off in say 10 seconds, then do a power on power off cycle to actually re-boot it.

Othewise the changes you made are not actually made in the unit.


Anyway, I have updated mine to the .55 version.


When you set up the trunk in pbxnsip, make sure it is set as a SIP Gateway.

Add the IP address of the Grandstream only in the Domain field.

Make CO Lines co1 co2 co3 co4 etc... I don't think that has anything to do with the operation of the lines.

The Password doesn't matter since we are not registering the gateway.


In the Grandstream there are several items to change.

In Profile 1 put the IP address of the pbxnsip in the SIP Server field

Also, change SIP Registaration to NO.

Thats it for that page.


On the Channels page:

Put a number (1,2,3,4,5,6,7,8) for each actuall working PSTN Line that is going into the gateway in the CHANNEL(s) Field.

Then set the Profile ID to 1 for working PSTN Lines ONLY. Set others (unused PSTN Lines) to Profile 3.

Thats it for that page.


On the FXO Lines page:

Set Wait for Dial Tone to NO for all channels

Set Stage Method to 1 for all channels

Set Min Delay Before Dial PSTN to 500 for all channels

Leave everything else on the page in default settings.


That is how I set mine up. And it did take about a day or so to figure it out since I was learning as I was doing it.


If you are in the USA and have toll free dialling you can call me. Email first.

If you set up a FREE SIP Account with WWW.Callcentric.com you can call me on 1 777 238-8032.


Bill Hayhurst

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Hi Bill. Thanks so much for your help.


Here is what I did.


I upgraded to the latest firmware. .55 i think?


I tried what you said, but it didn't work for me, so here is what I did.


I set the trunk as an outbound proxy. Set the Outbound Proxy to the IP of the gateway, set the redirection, and co1...co4.

On the gateway, I have Wait for dialtone set to yes, one stage dialing, and 100 s wait before dial. I can make inbound calls and it works great every time, but outbound calls are hit or miss. Sometimes it works, sometimes it doesn't. Do you have any suggestions with what I should try? I'm not sure, but I think the problem is that the outbound call doesn't always make it as far as the gateway.


I'm so confused!!


Thanks again, I really appreciate it.

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Hi again-


Writing this time to let you know that I finally got it to work!! I set the min delay to 750 and the current disconnect to 300 and presto! it works! Not sure why those settings work that way, but that's what grandstream told me to do, and it worked no problem! Thanks again for the help!

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Hi again-


Writing this time to let you know that I finally got it to work!! I set the min delay to 750 and the current disconnect to 300 and presto! it works! Not sure why those settings work that way, but that's what grandstream told me to do, and it worked no problem! Thanks again for the help!




I am happy to hear that it is working now. You can check your Firmeware Version on the Status page of the GWX-410x


The reason your unit would work hit and miss was due to time it takes the Dial Tone from the PSTN to begin after you go Off Hook. With your original setting of

100 (milliseconds not seconds) your unit began to dial before the PSTN was ready to accept any digits. There are many Registers in a PSTN where the Dial Tone originates from. Some are a little faster and some a little slower. You were on the edge with 100 ms..

I suggested a 500 millisecond (1/2 second) delay before dialling to remedy the situation. With your setting of 750 ms. you are even safer.

If you go too high, then the call will take longer to process after you finish dialling.


The Current Disconnect value of 300 means that if the PSTN sends a Break in the Loop (PSTN Line) for 300 ms. the Gateway will release its connection and the call will be disconnected. This is the Calling Party Control (CPC) in PSTN talk. (Similar to a SIP "BYE")


Well now you can try using and learning a Grandstream GWX-400x analog adapter.

In some ways it is similar to the GWX-410x and in other ways it is 100% the opposite.

With it, you will be able to add a cordless telephone and other non-SIP telephones or devices to your PBX.


Bill Hayhurst

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  • 4 weeks later...

Calls are going in and out fine, however, we're having some issues with call quality.


Whenever a person on our side of the pbx speaks, the opposite side is muted for about a half of a second followed by a half of a second of static when they stop speaking.


What could be causing this and how can I fix it?


The other party does not hear any of this, and they report that the quality of the call is fine.



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So I did some playing, and the Silence Supression doesn't seem to make a difference. Here's what I think that I have figured out. When echo cancellation is ON, it sounds the way I had described above. When it is OFF, the other party can be heard fine, but the person on our side of the PBX is echoed. Hmm...any thoughts?

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  • 3 years later...

I worked through this with my adapter and here is how I set mine up:


Grandstream POTS Adapter:


Product Model: GXW4104

Software Version: Program-- Loader-- Boot--


** Do the FXO line test on all the lines and check AC Impedance, CPT Detection, CID Detect. Check the apply test results automatically and to all ports (unless you are doing one at a time).


If your adapter is exposed to the internet change the password under advanced settings.




FXO Lines: - Channel Dialing To PSTN


  • 1. Wait for Dial-Tone(Y/N): ch1-4:N;
  • 2. Stage Method(1/2): ch1-4:1;
  • 3. Min Delay Before Dialing Out: ch1-4:500;


FXO Lines: - Channel Dialing to VoIP


  • User ID: ch1-4:1; ## Without a user ID the adapter would not call the pbx, it seems to work with any userid.
  • Sip Server: ch1-4:p1;
  • Sip Destination Port: ch1-4:5060;





  • Make sure all the channels you are using are set to profile 1.
  • I noticed the audio in was kind of low so upped the RX to 7. Beware that all POTS lines I have dealt with are different and will have to be tuned to your liking. Sometimes the ringback tone is too low and may need to be increased.


Profile 1

  • SIP Server: xxx.xxx.xxx.xxx (PBX IP)
  • SIP Registration: No




snomONE PBX:


  • Type: SIP Gateway
  • Direction: Inbound and Outbound
  • Trunk Destination: Generic Sip Server
  • State: Enabled
  • Display Name: Grandstream
  • Domain: xxx.xxx.xxx.xxx "IP of your gateway"
  • No User Name Or Password
  • Accept Redirect: Yes
  • Interpret SIP URI always as telephone number: Yes
  • Send Call To Extension: "The Extension You Use"


Make sure you remove the area code out of your domain settings so it does not dial 10 or 11 digits


Make sure to have a 7 digit dial plan active that points to the pots adapter.

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