Call problems between 2 webRTC phones and Asterisk 13.13

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.
connect 6005.
connect 6006
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

failed to get local SDP" back from

You seem to have a problem downstream.

Also, the upstream has sent an unresolveable conctact header, so Asterisk will not be able to initiate transactions properly in the reverse direction.

Please elaborate on the downstream problem? Any idea for a solution :slight_smile: ?

Ask the supplier of the downstream system what will cause the error message. 603 is declined, but the text with it makes me think there is a configuration problem.

Declined just indicates that the request was refused for policy reasons.