Firmware for UCM6301/6302/6304/6308/6300A/6302A/6304A/6308A released as Beta



Dear Grandstream Customers,

Firmware for UCM6301/6302/6304/6308/6300A/6302A/6304A/6308A is now released as Beta.
Download Release notes
Download Firmware

For UCM6300 with RemoteConnect, make sure to download the latest Wave applications (1.0.15.x/1.15.x release) from here.

Please contact us should you have any issues. Thank you for your support for Grandstream products.

Technical Support
Grandstream Networks, Inc.



I’m trying the new UCM630x FW beta, I immediately notice a couple of things:

  • Trunk WebRTC very useful (I had reported it a few months ago), but the icon in English “Contact Us” is created -> you need to be able to change this icon with the desired writing (we are not all English)

  • on the WebRTC Trunk the caller has an excellent choice of being able to choose microphone or speakerphone on the mobile phone, only that the icon of choice is too large so the caller from the mobile phone struggles to see and manage it
    And rather I would have set it by default as a microphone and not as a speakerphone.

  • "[System] EXT4 is now the recommended file system format for external storage instead of NTFS"
    in my opinion it is necessary to insert a key with which UCM63OX can autonomously format USB / SD Card
    in EXT4, not all installers are practical in doing this, moreover I hope NTFS support remains as with EXT4 there would obviously be problems in reading the contents of USB / SD Card
    in EXT4.

  • [Extensions] Resetting an extension will now delete all IM-related data for that extension.
    what does “Resetting an extension” mean and how to do it

  • MTU max 1492, what’s the point of not being able to put 1500 as a local LAN port?

  • the link below shows port 1234, it is better to change the example to 5061 as it has been unified and it is always this.


  • [Wave] Optimized process when calling extensions on the private local network.
    I was hoping the optimization was:
    in local network WAVE automatically uses the local network, so it registers for example on
    instead on the data network it passes on the public ip of mobile data.
    Instead, locally it continues to use the router’s Public IP, so going from the internet even local calls use internet bandwidth, I hope it can be solved.

  • regarding the login on WAVE mobile via QRCode it works and is excellent, as far as WAVE Desktop is concerned it would be better if the email sends a unique link, perhaps in time, which would be very convenient for the Customer to click the link to log in.

  • unfortunately the Follow-me service problem continues to be a problem and makes the Follow-me service unusable. I have already reported it on several occasions with tests, screens and videos via tickets and directly to you at least 1 year and a half ago until today. :frowning:

  • it would be advisable to have WAVE mobile multi-account, for example I want to register 210 WAVE mobile on the local network and 210 WAVE mobile on Remote connect to use the 210 remotely, today it is not possible.
    In reality it would be very useful to make a single registration on 210 where WAVE sets up both the local server and the remote connect server, so that the customer will not have to do anything in case he leaves or enters the company.

  • I did further tests and the audio on Trunk WebRTC is really very poor, even forcing any codec, I want to clarify that I am testing on fiber 300/300 and I had no traffic in progress.
    First I tried with the mobile phone on local wifi and the audio was satisfactory, obviously this is not the goal

  • in my view, the visualization of the WebRTC Trunk situation on the dashboard (see screen 1) is to be improved as the “dot” remains gray even if the Trunk is active and functioning, while on the WebRTC Trunk the “dot” is green if active, in my opinion it would be enough to align the “green” color also in the dashboard.



  • using some codecs, for example GSM and G729 on the webrtc trunk you are put into ban, it would be useful to understand which codecs cannot be used.


Tried the beta but Just had to install the previous firmware, main issue was that when we dial out the trunk supplier account number was showing on the phone, no idea why but is now showing the correct number dialed on the previous firmware…


The WEBRTC TRUNK is fantastic… Loaded on my office system and was able to get users on my web site to access the trunk which routed to my call queue. GREAT WORK!


We are having the same issue on all registered SIP TRUNKS. Doesn’t seem to happen on PEER SIP trunks, but still testing. Since this is on our office system, we are living with it for the moment, however, we couldnt use this firmware on a customer’s system as it current stands.


Had a good play with WEBRTC today, seems to work perfectly with Edge, Chrome and Opera but try to initiate a call from Firefox and nothing happens.


to my knowledge Firefox does not support WebRTC, the other browsers you mentioned are all Chromium based


Many thanks Damiano 70, that would explain it then. Seems a shame really because I could not implement it into my websites knowing a lot of my customers use Firefox.


I think Firefox supports it partially, I’m not a computer scientist, but I think you can make the link open with a certain browser.



Thank you all for the feedback!

Yes, NTFS will still be supported so no need to worry.

You can reset an extension by clicking on the More button in the Extension overview page. Resetting an extension will reset all settings to default settings except for the concurrent registrations setting, delete all IM-related data and recordings, and remove the extension from any scheduled meetings.

Development is still working on a more convenient method to log into the desktop version.

We apologize for the delay in addressing this issue. I will continue to be following up internally with this problem.

Could you provide details about the environment where audio quality is poor? It doesn’t seem like others here are experiencing this.

I’m not sure what you mean here. Could you explain this?

I will report this issue and all of the other issues brought up in the replies internally.


ok, I thought it was a new features code example * 61 that resets the extension settings (example removes deviations, DND and …) when done by the phone, useful for messy customers. Hope this is on the agenda.

Could you provide details about the environment where audio quality is poor? It doesn’t seem like others here are experiencing this.
I dealt better with this problem, practically the same conditions using my Realme GT Master edition phone the audio is discreet, if I use Motorola Moto E4 Plus XT1771 for example the audio is very bad.
Probably WebRTC uses a lot of the phone’s resources, and Motorola Moto E4 Plus XT1771 having few resources cannot manage the WebRTC correctly, maybe it is possible to improve the resource management?
I imagine a scenario where customers can have “new” and less new cell phones, and in the second case it would become difficult to explain the technical problem.

I’m not sure what you mean here. Could you explain this?
if I set the GSM codec on the WebRTC Trunk, for example, in addition to not working (if I call nothing happens), by repeating the call my public ip goes into ban (see screen), in the drop-down menu of the codecs that can be used by WebRTC you should only enter the list of codecs that can actually be used (to me it is PCMU / PCMA / OPUS)


Importing extensions isn’t working properly on this. I literally exported the extensions and tried to reimport them and its telling me there are a bunch of issues with the config excel file when I didn’t even make changes to it. There was no problem on the previous firmware with this.


the new fw beta has several more fields in the extension, I think that is the error, just export the new example file and copy the desired fields to it.


going to give this a try , the previous SDK file method did not work well


Thanks for the clarification. Just for confirmation, are you talking about calling from your phone’s browser, or are you talking about receiving webrtc trunk calls? If the latter, are the calls routed to Wave app on your mobile phone, or are the calls doing directly to your mobile phone number?

While I understand that MTU value of 1500 for LAN is the standard value set by IEEE802.3, could you please explain how this affects your environment/deployment?


Can the Mobile Wave App support more than one account please like Wave Lite did.

Very handy for us to have an account on all the systems we support to check functionality.


I mean use the WebRTC trunk from the mobile browser (Chrome updated)


it does not affect, but since any device in the local network has MTU = 1500, I thought it was an error / forgetfulness


I just tried this on my 6302A with Exported SIP extensions, then imported back to the same system (set the DUPLICATE action to UPDATE)

It was successful. I will try this later on my 6302. Can you give some more info on exactly what you were trying?