I have TLS and SRTP working for normal calls on an extension. However when connecting to a conference internally there is no audio (cannot hear welcome message). When the extension is changed back to UDP and RTP then no issues in hearing the conference welcome message.
Is there a setting on the conference or somewhere else to allow both TLS and UDP to work?
I have also changed the standard ports away from 5060 and 5061. This is for SIP CHAN. With the ports changed it still works fine with UDP but no luck with TLS/SRTP. FreePBX 18.104.22.168.
Appreciate advice on this.