[SR-Users] ACK and Bye not relaying when record_route is not used

Issue : Not getting relay of ACK and BYE to the next hop after the call is answered my Scenario : Asterisk ------->kamailio sip proxy-------------------> carrier (outgoing call) My carrier is not allowed to send the SIP packet with Record-Route header. So that I have removed record_route(). After that the call is getting connected. I am getting 200 OK (SDP) from carrier side and forwarded that to the Asterisk on the other side. As a response I am getting ACK from asterisk. But the kamailio is not forwarding the ACK to the carrier side. I understood this is because the record-route is not there. The same thing is happening for BYE also. The Bye is not forwarding to carrier side. Kindly suggest me a solution for this for relaying ACK and bye without Record-Route in kamailio Bellow is the 200 OK SDP I am sending back to asterisk 2024/06/02 10:27:04.756610 103.155.114.101:5060 -> 103.182.153.113:5060 SIP/2.0 200 OK Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 Via: SIP/2.0/UDP 103.182.153.113:5060 ;received=103.182.153.113;rport=5060;branch=z9hG4bKPj463fedcf-6258-4655-a53d-58ea57f144af To: <sip:09496381412@103.155.114.101 From: <sip:917946357720@gaesip.teleforce.in CSeq: 22823 INVITE Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE Contact: <sip:lucentNGFS-115804@10.5.110.117:5060 ;alias=10.5.110.117~5060~1;x-afi=11> Content-Type: application/sdp Session-Expires: 7200;refresher=uas Supported: timer Content-Length: 248 v=0 o=LucentPCSF 130227946 130227946 IN IP4 103.155.114.101 s=- c=IN IP4 103.155.114.101 t=0 0 m=audio 12806 RTP/AVP 8 101 ------------------------------------------------------------------------------------------------------- The ACK I am getting back from asterisk is 2024/06/02 10:27:04.760392 103.182.153.113:5060 -> 103.155.114.101:5060 ACK sip:lucentNGFS-115804@103.155.114.101:5060;alias=10.5.110.117~5060~1;x-afi=11 SIP/2.0 Via: SIP/2.0/UDP 103.182.153.113:5060 ;rport;branch=z9hG4bKPjb7299c25-13d3-484e-8e78-e7c83620edce From: <sip:917946357720@gaesip.teleforce.in To: <sip:09496381412@103.155.114.101 Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 CSeq: 22823 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 18.13.0 Content-Length: 0 Thanks Arun