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 ->
103.182.153.113:5060SIP/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>;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109
From: <
sip:917946357720@gaesip.teleforce.in>;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1
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 ->
103.155.114.101:5060ACK 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>;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1
To: <
sip:09496381412@103.155.114.101>;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109
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