As mentioned above - use TOPOS BUT - return record_route() back. On Sun, Jun 2, 2024, 08:40 Nick Digalakis via sr-users < [email protected]> wrote:
> Hi! > > Check out the TOPOS module. > > It does topology hiding and it should do what you want. > > On Jun 2, 2024 08:13, Arun K R via sr-users <[email protected]> > wrote: > > 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:[email protected] > >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 > From: <sip:[email protected] > >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 > CSeq: 22823 INVITE > Allow: > INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE > Contact: <sip:[email protected]: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:[email protected]: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:[email protected] > >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 > To: <sip:[email protected] > >;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 > > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to [email protected] > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: >
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