I have FreeSwitch working with SIP Clients for
- Extension to Extension Call
- Extension to PSTN / Gateway Call
- PSTN/DID to Extension Call
I have configured WebRTC with SIPML5 clients and it is working on following scenarios
- Extension to Extension Call
for Extension to PSTN/ Gateway FreeSwitch routes call to TRUNK but its not connected.
How can I get it working ?? What is missing parameter?? I think something at bridge is required to
FS Console logs are available here