Hi, We are trying to replace our Adtran which is being used as an SBC for faxing with openSIPS. But we are getting a problem with openSIPS. This gonna be our call flow ->
Asterisk -> OpenSIPS -> SIP Trunk Rtpproxy is running inside openSIPS. I am new to openSIPS so need help from smart folks. When we get 200 OK (SDP) from trunk provider, our ACK is not traversing the openSIPS. ACK is being directly sent to trunk provider. Please view the call flow -> 2019/09/16 15:05:16.045125 10.250.110.90:5060 -> 10.10.10.172:5060 INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.250.110.90:5060;branch=z9hG4bK5b901208 Max-Forwards: 70 From: "Sunny Singh" <sip:[email protected]>;tag=as6597b08e To: <sip:[email protected]> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX certified/13.21-cert3 Date: Mon, 16 Sep 2019 20:05:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 301 v=0 o=root 736228850 736228850 IN IP4 10.250.110.90 s=Asterisk PBX certified/13.21-cert3 c=IN IP4 10.250.110.90 t=0 0 m=audio 12276 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv 2019/09/16 15:05:16.045914 10.10.10.172:5060 -> 67.231.12.67:5060 INVITE sip:[email protected]:5060 SIP/2.0 Record-Route: <sip:38.65.55.146:5060;lr;ftag=as6597b08e;did=455.89ce1c5> Via: SIP/2.0/UDP 38.65.55.146:5060;branch=z9hG4bK424b.1d038492.0 Via: SIP/2.0/UDP 10.250.110.90:5060;branch=z9hG4bK5b901208 Max-Forwards: 69 From: "Sunny Singh" <sip:[email protected]>;tag=as6597b08e To: <sip:[email protected]> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX certified/13.21-cert3 Date: Mon, 16 Sep 2019 20:05:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 317 v=0 o=root 736228850 736228850 IN IP4 38.65.55.146 s=Asterisk PBX certified/13.21-cert3 c=IN IP4 38.65.55.146 t=0 0 m=audio 12322 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv a=nortpproxy:yes 2019/09/16 15:05:16.281867 67.231.12.67:5060 -> 10.10.10.172:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 38.65.55.146:5060;branch=z9hG4bK424b.1d038492.0 Via: SIP/2.0/UDP 10.250.110.90:5060;branch=z9hG4bK5b901208 From: "Sunny Singh" <sip:[email protected]>;tag=as6597b08e To: <sip:[email protected]>;tag=gK00b184ef Call-ID: [email protected]:5060 CSeq: 102 INVITE Record-Route: <sip:38.65.55.146:5060;lr;ftag=as6597b08e;did=455.89ce1c5> Accept: application/sdp Contact: <sip:[email protected]:5060> Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS Supported: replaces Content-Length: 236 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 687409 672900 IN IP4 67.231.12.67 s=SIP Media Capabilities c=IN IP4 67.231.12.207 t=0 0 m=audio 36368 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 2019/09/16 15:05:16.289521 10.10.10.172:5060 -> 10.250.110.90:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.250.110.90:5060;branch=z9hG4bK5b901208 From: "Sunny Singh" <sip:[email protected]>;tag=as6597b08e To: <sip:[email protected]>;tag=gK00b184ef Call-ID: [email protected]:5060 CSeq: 102 INVITE Record-Route: <sip:38.65.55.146:5060;lr;ftag=as6597b08e;did=455.89ce1c5> Accept: application/sdp Contact: <sip:[email protected]:5060> Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS Supported: replaces Content-Length: 253 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 687409 672900 IN IP4 38.65.55.146 s=SIP Media Capabilities c=IN IP4 38.65.55.146 t=0 0 m=audio 13406 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 a=nortpproxy:yes 2019/09/16 15:05:16.290712 10.250.110.90:34022 -> 10.10.10.172:5060 ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.250.110.90:5060;branch=z9hG4bK5c3259ad Route: <sip:38.65.55.146:5060;lr;ftag=as6597b08e;did=455.89ce1c5> Max-Forwards: 70 From: "Sunny Singh" <sip:[email protected]>;tag=as6597b08e To: <sip:[email protected]>;tag=gK00b184ef Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 ACK User-Agent: Asterisk PBX certified/13.21-cert3 Content-Length: 0 And for this setup we have to use openSIPS residential script or trunking script ? Thanks
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