Call problems between 2 webRTC phones and Asterisk 13.13

Asterisk Asterisk Support
1

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

2

failed to get local SDP" back from 192.168.1.120:58863

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.

3

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

4

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.