23 Mar
2016
23 Mar
'16
1:46 p.m.
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. -- Thanks, Rutu Patel <http://www.inextrix.com>