[SR-Users] t_check_trans failed on ACK

I am using Kamailio as registrar and proxy for underlying asterisk server. I am able to call but ACK on 200 OK respond from client phones are not validating transaction and thus not relaying correctly. Below are my Invites and 200 OK responses. Invite from Asterisk to Kamailio INVITE sip:31313004@rigel.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 From: "R31313005 N31313005" <sip:31313005@65.20.69.72:5060 ;user=phone>;tag=GR52RWG346-34 To: &quot;31313004(a)rigel.com&quot; <sip:31313004@rigel.com:5060> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 CSeq: 1 INVITE Contact: <sip:65.20.69.72:5060> max-forwards: 70 Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE Content-Type: application/sdp Content-Length: 228 v=0 o=Clarent 155866 155867 IN IP4 65.20.69.72 s=Clarent CallManager c=IN IP4 65.20.69.72 t=0 0 m=audio 45168 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Invite from Kamailio to Phone INVITE sip:31313004@65.20.67.118;lhst=180.211.119.50:25759;lm=midreg SIP/2.0 Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> Via: SIP/2.0/UDP 65.20.67.118;branch=z9hG4bK6cc3.dad7580d3e28f55cbb749dbb82f0c9eb.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;received=65.20.69.72;rport=5060;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 From: "R31313005 N31313005" <sip:31313005@65.20.69.72:5060 ;user=phone>;tag=GR52RWG346-34 To: &quot;31313004(a)rigel.com&quot; <sip:31313004@rigel.com:5060> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 CSeq: 1 INVITE Contact: <sip:65.20.69.72:5060> max-forwards: 69 Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE Content-Type: application/sdp Content-Length: 228 v=0 o=Clarent 155866 155867 IN IP4 65.20.69.72 s=Clarent CallManager c=IN IP4 65.20.69.72 t=0 0 m=audio 45168 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 200 OK from Phone to Kamailio SIP/2.0 200 OK Via: SIP/2.0/UDP 65.20.67.118;received=65.20.67.118;branch=z9hG4bK6cc3.dad7580d3e28f55cbb749dbb82f0c9eb.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport=5060;received=65.20.69.72;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 From: "R31313005 N31313005" <sip:31313005@65.20.69.72 ;user=phone>;tag=GR52RWG346-34 To: &quot;31313004(a)rigel.com&quot; <sip:31313004@rigel.com CSeq: 1 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: <sip:31313004@180.211.119.50:57935;ob> Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 322 v=0 o=- 3860150297 3860150298 IN IP4 180.211.119.50 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4016 RTP/AVP 0 101 c=IN IP4 180.211.119.50 b=TIAS:64000 a=rtcp:4017 IN IP4 180.211.119.50 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1461220511 cname:3b657c275d2a43f6 200 OK from kamailio to asterisk SIP/2.0 200 OK Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport=5060;received=65.20.69.72;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 From: "R31313005 N31313005" <sip:31313005@65.20.69.72 ;user=phone>;tag=GR52RWG346-34 To: &quot;31313004(a)rigel.com&quot; <sip:31313004@rigel.com CSeq: 1 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: <sip:31313004@180.211.119.50:57935;ob> Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 322 v=0 o=- 3860150297 3860150298 IN IP4 180.211.119.50 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4016 RTP/AVP 0 101 c=IN IP4 180.211.119.50 b=TIAS:64000 a=rtcp:4017 IN IP4 180.211.119.50 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1461220511 cname:3b657c275d2a43f6 ACK from phone to Kamailio ACK sip:31313004@180.211.119.50:57935;ob SIP/2.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport;branch=z9hG4bK111270de2729ade9e048afa0f9fbb8df Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> From: "R31313005 N31313005" <sip:31313005@65.20.69.72:5060 ;user=phone>;tag=GR52RWG346-34 To: &quot;31313004(a)rigel.com&quot; <sip:31313004@rigel.com Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 CSeq: 1 ACK Contact: <sip:65.20.69.72:5060> max-forwards: 70 Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE This ACK is failing on transaction check and not routing correctly to phone. Any help? Thanks