Asterisk 15 and Queuemetrics with WebRTC phone...

[cursor]

     I followed the instructions to set up Queuemetrics to use the integrated WebRTC phone.  Everything is running SSL (Lets Encrypt).  The webphone seems to register but becomes unreachable after a minute or so.  If I dial a number it takes about 30 seconds before I see Asterisk receive the call.  I get no audio on the call no matter is it is MoH or another extension.  The guide was made for regular chan_sip and not for PJSIP so I was wondering if anyone has been able to get the webphone working with Asterisk 13 or 15 and PJSIP.

     I have tried other webrtc webphones with the same configuration and they work with audio both ways.  This is all on a local LAN with no NAT at the moment although JsSIP and SIP5ML both work from outside the local network.  Any pointers?

[Itgigi (Loway)]

Hello cursor,

Thank you for your post, I've notified the developers about the issue you've encountered and posted a bug report (ref. #3830).

It would be very helpful if you could attach as much information about your configuration as possible to help us debug the issue further.

You will be notified once we have more information

Best regards,
Iacob