Outgoing Call "Not Found" but call is success

[Ravhi Rizaldi]

Hi, I don't know where the problem is but JSSIP cause Call Not Found but the call is success! 

Here is my logs :

as you can see the call is accepted :

INVITE sip:900999085...@172.20.30.61:8089 SIP/2.0
Via: SIP/2.0/WSS 172.20.30.61;branch=z9hG4bK3920468
Max-Forwards: 69
To: <sip:900999085...@172.20.30.61:8089>
From: "1080" <sip:10...@172.20.30.61:8089>;tag=lq69p82207
Call-ID: jsg7fcm3sttqr41ccalq
CSeq: 2571 INVITE
Contact: <sip:10...@172.20.30.61:8089;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: Kalapa SIP Client
Content-Length: 6261

SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS 172.20.30.61;branch=z9hG4bK7868502;received=172.10.10.89;rport=63124
From: "1080" <sip:10...@172.20.30.61:8089>;tag=lq69p82207
To: <sip:900999085...@172.20.30.61:8089>;tag=as52db16be
Call-ID: jsg7fcm3sttqr41ccalq
CSeq: 2572 INVITE
Server: Asterisk PBX 16.26.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:900999085...@172.20.30.61:5060;transport=ws>
Content-Type: application/sdp
Require: timer
Content-Length: 980

Any ideas to fix this? I'm using sipml5 to test my backend it's fine! call is made and no error.