Hi,

We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server.
 
Call flow is as below:

User => opensips => FS => PBX,

but call not sending out from FS and drop by FS.

Here I am attaching both sip logs and freeswitch logs.

Can anyone please help to sort out the issue.