Hi, IZ ME AGAIN. Sorry for 2 questions in a day.
I managed to get webRTC phones to connect to Asterisk through a secure websocket. The problem is when I try to make a call between them.
call 6006 from 6005
(Asterisk outputs phone is busy.It shouldn’t be. I just connected it via sipml5)
Here’s the output from Asterisk with high verbosity : here
Sorry for not being able to copy pasta i’m just plain retarded with centos and don’t know how to copy and I typed the output.
They don’t always disconnect from server. (They this time did.) But I can neither pick up from the called phone nor can I HangUp or hangup the call from the callee. But I can see the called phone ringing ringing through the SipML5 interface. I just can’t do anything.
Here’s the sip.conf for the phones and extensions.conf for the context group if needed : Here