I’m super excited to share that the new Sangoma Phone Browser-Based Client is now available to all Sangoma Phone softphone license holders at no additional cost. This new option runs entirely in your web browser- no installation needed and delivers the exact same feature set as the desktop app.
Already a softphones subscriber? Try it now and share your thoughts
It works, but I want to host it on the PBX, since I have to allow access to the PBX anyway for softphone users.
If it’s hosted on the PBX, then it can also pre-fill the hostname and port 6443 stuff so that users can log in just with their username and password (and MFA, if configured), just like using UCP.
And speaking of that, once it’s hosted on the PBX, it can be made into a UCP widget and the kudgy old webrtc phone that is there now can go away and we will have a very nice integrated solution.
Thanks for the feedback – Just a quick note. This supports MFA (I’m sure you’ve noticed) and we’ve created a link so that you can jump right in from the new UCP sidebar widget.
This assumes that your PBX has Sangoma Softphone licensing that is set up and working properly. If that’s the case, you’ll need to include the RTAPI port when identifying your PBX Example:https://mypbx.local:6443
Does this only work if your system is open on the internet for access, or can it work over a local connection via IP?
When I am trying via a local IP, I am getting a “Lost connection to PBX. Reloading…” on the screen.
You will need access to the internet - In this case, we see that error when the port is not defined. Do you have the port switch :6443 added to your host?
Locally, yes. We only use the soft client locally or over VPN, we don’t leave port 6443 open to the internet.
Does this web phone connect from only certain IP addresses or domains that we can put into an ACL on the Firewall?
If we could get IPs that we could put onto an ACL, this would be very useful for us.
Hi @Drakulahn There is no need to whitelist any specific IPs in the FreePBX firewall for the browser phone to work. The browser simply needs to be able to reach the Freepbx RTAPI port (default 6443) and the RTP media ports.
However, when connecting the phone using an IP address instead of a domain name, the browser may block the WebSocket connection due to certificate warnings. To work around this, open a new browser tab and manually visit: https://<PBX_IP>:<RTAPI_PORT>
Then click “Accept and Continue” to bypass the certificate warning. Once that’s done, the browser phone should connect without issues.
So I am able to connect to my extension through the browser now as I opened port 6443.
My Sangoma Talk app works just fine.
But I am not able to get any audio to work through the browser.
Same when I am remote with the desktop client without a VPN, I cannot get any audio to work.
I have the RTP ports open, and my Sangoma Talk app works just fine.
Does the Desktop app direct RTP audio a different way than the Sangoma Talk app does?
I have had some systems where I have had to put a Stun server in the special settings under each user to get the audio to work right for Sangoma Talk.
Hi We have been testing the web client and have been very happy, except one issue that has come to light recently
If we close the computer down it is now losing the address on teh login so need re entering , but does keep user and password amd also its losing the popup url , , both of these are a bit of a blocker in deploying to customers to replace teh ‘App’
FYI this is on Macs with chrome Version 139.0.7258.68 (Official Build) (x86_64)
It didnt initally lose the PBX url on teh login , that seems to be a issue in teh last few weeks
HI @cyberco
Can you please upgrade the Chrome browser to the latest version – 138.0.7258.139?
We have verified that with this version, the hostname and username are retained (if the “Remember extension/username” checkbox is enabled) even after a multiple reboot or power off, and upon launching Chrome again.
Hi Thanks for the Reply, Yes Im running that version of chrome Version 139.0.7258.139 (Official Build) (x86_64) and phone is v4.1.0 but still the same issue.
Extension and password are retained but the Host is Blank and so is the pop url and any other settings when I open Chrome again.
Got it working with filling in the ICE Host Candidates and the WebRTC Settings > STUN Server Address under Asterisk SIP Settings.
Did my NAT settings for the ICE Host Candidates and used a google STN server for the WebRTC settings, and now my Audio is flowing correctly when I connect outside my network.