SIP Call (Busy) Error

[sangsoo]

Hello.
I'm testing SIP functionality, and when I make a call, it returns "Busy."
- https://docs.flashphoner.com/display/WCS52EN/A+call+between+two+browsers+made+via+the+SIP+server
- The "Without an external SIP server. SIP and RTP media are processed by WCS" section in the documentation works fine.

I configured andrius/asterisk as a Docker image on a different server (not WCS) and ran it in network-hosted mode.
- https://hub.docker.com/r/andrius/asterisk

My wcs flashphoner.properties:
dtmf=RFC2833

-----Version info-----
wcs_version=5.3.126-4162ea876dc627c303f32b785af69ac5304261c5
wcs_client_version=2.0.259-485b3fbffa4fd4abe88fbdc455691a05bbf79025

I'm trying to connect a call using two extension(internal) numbers (7003 and 7006).
The registration succeeds, but when I answer the call, it immediately disconnects with a 486 error message.

I've attached the asterisk log (486.txt). Can you help me figure out what the problem is?

Also, what configurations are required for the next two steps in the Operation flowchart? I'd like to check each flow.
1. SIP server as a proxy server to transfer calls and RTP media
2. SIP server as a server to transfer calls only

Please answer. Thank you.

 

[Max]

Good day.
Please update WCS to 5.3.165 and test again. The issue may be in DTLS handshake on callee browser side.
Please also note that actual 5.3 docs are here: https://docs.flashphoner.com/static/WCS53/

 

[sangsoo]

Thank you for your reply.

Referring to the article, I updated to version 5.3.165, but I'm still experiencing the same issue (486 busy).
I didn't install a separate certificate for Asterisk. Could this be related?
I've attached the logs for WCS and Asterisk. Is there anything else you'd like to see?

Thank you.

 

[Max]

Seems like audio codecs negotiation does not pass:

Please try to set the following WCS option

allow_outside_codecs=false

if this does not help, please collect a full report and traffic dump on the server side. Send the report using this form.