Hello, Thank you for your response, Could you please tell me what do you mean by return record_route() back ? I have implemented topos module and now call is working with record_route(). But now for the BYE I am getting 403 GLU unmatched from the carrier. Could you suggest something for this
Thanks again Arun On Sun, Jun 2, 2024 at 5:36 PM Yuriy G <[email protected]> wrote: > 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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