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Update 6 BETA - Try out our new web based softphone!

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ok thanks just trying to test a template i put together and needed http is it on hte same port 5001 or uses 5000?

thanks leon
 
Opus codec has bugs. This is why we did not add it on all the phones. When Yealink team are finished with the testing of the codec, we will add the codec to the phones, and you will be informed via the change log. Feel free to add it yourself and report issues you find.

thanks for your response nb, Could you explain how to add it manually? on Yealink side i have added it, on 3cx side i want to add and try it. Right now added on yealink side and have some minor noise sometimes in the sound..

Glad to know it will be available on all yealink series soon.
 
I am trying out the new web client and i get this error when trying to answer a call from a video door phone.
upload_2018-8-14_16-10-4.png
I can make a video call to the door phone with the new web client, I just cant receive one with it.
is there a setting in need to change somewhere for this to work?
Thanks.
 
I am trying out the new web client and i get this error when trying to answer a call from a video door phone.
View attachment 8305
I can make a video call to the door phone with the new web client, I just cant receive one with it.
is there a setting in need to change somewhere for this to work?
Thanks.

This commonly happens when the destination you call has SRTP enabled.
 
Will a call between users on bridged systems be able to upscale to be a video call in the future, or should it work already? I have done the usual firewall checks etc.
 
This commonly happens when the destination you call has SRTP enabled.

The client I was calling was the 3cx web client which is why I posted the message. is there a way around this or not? Just trying to help with finding issues and letting 3CX know as this will be a problem for anyone wanting to use a video door phone and the new web client phone.
 
I still FAIL to see why you chose to give the logo to enterprise ONLY. I am an end user and management would be very happy to have this on our phones. Paying to have a logo on our phones (Upgrade to enterprise) is about the only real feature we would get from enterprise.

Please just enable it for PRO. I completely understand for the FREE version, but we can and do pay a license fee, and maintenance EVERY YEAR, just enable it for PRO.

Also since this upgrade we now get "Waiting for network" and eventually register failed, whereas pre-upgrade no issues. we are all android phone owners

why not just change the graphic file on the server? sure, you might have to copy them back in after an update - but pretty cheap option.
 
Good work!

I like the new features from the new Update 6 BETA!
 
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Will a call between users on bridged systems be able to upscale to be a video call in the future, or should it work already? I have done the usual firewall checks etc.
@rdcomms, Yes video call work on bridge seamlessly.
 
1. The webclient is a bit slow for call setup on RTP (maybe ~1 second or so) is this something that would be improved?
2. Would it be possible to get some kind of info or statistics button on the client to show call info such as rtt/jitter/codec in use?
3. How does the phone provisioning area on the webadmin affect the web client, should a user relogin to apply changes made? we're noticing some inconsistencies with the codec orders in SDP when changing these.
 
1. The webclient is a bit slow for call setup on RTP (maybe ~1 second or so) is this something that would be improved?
2. Would it be possible to get some kind of info or statistics button on the client to show call info such as rtt/jitter/codec in use?
3. How does the phone provisioning area on the webadmin affect the web client, should a user relogin to apply changes made? we're noticing some inconsistencies with the codec orders in SDP when changing these.

thanks for the feedback.
1 sec is acceptable I think. When you think that your phone that raises the whole audio stack is a browser. Sometimes it takes much longer to load a simple webpage.. Remember you have other tabs open.
2 you can use chrome://webrtc-internals. Maybe we will make our own soon as it is inevitable for troubleshooting but for now, use the best tool out there. If you open chrome://webrtc-internals while in an active WebRTC session, you will immediately see the API trace, events and all connections.
3 Webrtc is a library that for every single thing you change, you need to refresh. Change microphone? refresh. Change something whatever it is - A refresh is required. And if you are changing stuff related especially to codecs, then the audio audio stack needs to be revived. Also not all codecs are used for Webrtc.
 
I'm not sure how it could be integrated.

But any chance the JabraDirect software can be linked to the WebClient? Our clients enjoy using the hardware buttons on the headsets to answer the SoftPhone at present.

It is already linked - plugin a jabra and try it
 
ok thanks just trying to test a template i put together and needed http is it on hte same port 5001 or uses 5000?

thanks leon

5000 is the http port. Look at other legacy templates and copy there. Remember that these tips move you in the direction of having unsupported templates and unsupported phones.
 
thanks for your response nb, Could you explain how to add it manually? on Yealink side i have added it, on 3cx side i want to add and try it. Right now added on yealink side and have some minor noise sometimes in the sound..

Glad to know it will be available on all yealink series soon.

That's exactly the problem. There is some problem we are dealing with Yealink on this one. There are a number of OPUS codec versions. There are a number of OPUS codec variants Sampling from 8 kHz (narrowband) to 48 kHz (fullband) so its not like the other codecs, where you select one and it is universal. We are working on this with yealink and we will release template updates soon.
 
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right i wanted to test something but I have decided to not do it since time is premium and I am making sure those devices are removed. Thanks leon
 
not sure if this is a beta glitch or not but I have tried setting up a new Call Queues group and i can't get one of the phones in the group to ring? The same phone will ring in any of the old Call Queues I already had on the system but not in the new one.
Has any one else had similar issues?
 
not sure if this is a beta glitch or not but I have tried setting up a new Call Queues group and i can't get one of the phones in the group to ring? The same phone will ring in any of the old Call Queues I already had on the system but not in the new one.
Has any one else had similar issues?
Which Version are you using? If the beta6 which include the new web softphone in the web client, then please wait for final release if you are experiencing issues. Note that any beta is advised not to use in production system.

Make sure if you have added the agent required to be in your new queue first then try again.
If still same, click on "services" in the management console Dashboard, find"3CX PhoneSystem 01 Queue Manager Server" then restart it. and try a test once it has been restarted successfully.

Let us know if issue still here.
Thanks
 
Which Version are you using? If the beta6 which include the new web softphone in the web client, then please wait for final release if you are experiencing issues. Note that any beta is advised not to use in production system.

Make sure if you have added the agent required to be in your new queue first then try again.
If still same, click on "services" in the management console Dashboard, find"3CX PhoneSystem 01 Queue Manager Server" then restart it. and try a test once it has been restarted successfully.

Let us know if issue still here.
Thanks
It's beta 6 and is on my company's in house test system.
Yes i have tried re-starting the services but this has not fixed it.
I'm fine to wait for the final build and see if that sorts it but was just giving you a heads up in case there is a issue on the beta build that needs fixing before the official build is released.
 
It's beta 6 and is on my company's in house test system.
Yes i have tried re-starting the services but this has not fixed it.
I'm fine to wait for the final build and see if that sorts it but was just giving you a heads up in case there is a issue on the beta build that needs fixing before the official build is released.
Hi, The final built is here and you can update to latest version. The only thing is OPUS codec still not yet available for yealink phone t2, t4 etc.
 
Can someone tell me when OPUS codec will be available for Yealink T2, T4 series etc? It only available on the phone if you set it manually but not on 3cx side.

Thanks
 
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