We have a call flow where the agent needs to transfer the call back to the PSTN. We have an agent application that sends the REFER to the sipX server, but instead of the sipX server forwarding that REFER to our SBC, it is generates an INVITE and sends it to our SBC.
The call completion works since our SBC sends the call to our PSTN gateway, but since it's an INVITE the sipX server has to be in the call leg for the entire call duration. We want the sipX server to forward the REFER to our SBC so sipX can get completely out of the call flow. I do have our SBC configured as an unmanaged gateway, and I have a calling plan that uses our SBC for the route. I'm not sure if I need to be doing something with sip trunking or do we need to format our REFER packet differently? I know sipX supports the REFER since it's accepting it with a "202 accepted" response. Ideally the sipX server should send the REFER back to the same endpoint that originated the call to the ACD, but it doesn't seem to do that on it's own. thanks, James 6000 = ACD line 10.103.4.46 = sipX server 10.1.32.132 = agent 10.103.237.42 = SBC 10.103.237.224 = original caller source (sip proxy/gateway) 8005250280 - call completion number This is REFER from Agent client to sipx REFER sip:[email protected]:5150;LINEID=a2c75c002919221d2d20739304f6c118;x-sipX-nonat SIP/2.0 Via: SIP/2.0/UDP 10.1.32.132:5074;rport;branch=z9hG4bK55151 Route: <sip:10.103.4.46:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMTZjNDMyMTk4MDc4%2114f9304e82467cf973005ce29dc1857f> Max-Forwards: 70 To: <sip:[email protected]>;tag=16c432198078 From: "Queue1" <sip:[email protected];sipx-noroute=VoiceMail>;tag=0c57f4af4611f9a6 Call-ID: [email protected] CSeq: 1 REFER Contact: <sip:[email protected]:5074> Expires: 3600 User-Agent: mjsip stack 1.6 Allow: INVITE, ACK, CANCEL, BYE, INFO, OPTION, REGISTER, UPDATE, SUBSCRIBE, NOTIFY, MESSAGE, REFER, PUBLISH Refer-To: <sip:[email protected]> Referred-By: "Queue1" <sip:[email protected];sipx-noroute=VoiceMail> Content-Length: 0 INVITE sip:[email protected] SIP/2.0 Route: <sip:10.103.237.42;lr> Record-Route: <sip:10.103.4.46:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMTZjMTY2NzMxNjU2NA%60%60%214d83309cc152a56bf4ff190922be777d> X-Sipx-Authidentity: <sip:[email protected];signature=4A0ACE91%3A%3A8ed4d3ccaa2968128384d6ef663fa66c> From: <sip:[email protected]>;tag=16c1667316564 To: sip:[email protected] Call-Id: s-42a2214f969f9ea8-268 Cseq: 1 INVITE Contact: <sip:[email protected]:5150;LINEID=a2c75c002919221d2d20739304f6c118;x-sipX-nonat> Content-Type: application/sdp Content-Length: 198 Referred-By: "Queue1" <sip:[email protected];sipx-noroute=VoiceMail> References: [email protected];rel=refer Date: Wed, 13 May 2009 13:43:45 GMT Max-Forwards: 15 User-Agent: sipXacd (4.0.0-015321) (Linux) Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, INFO, REGISTER, SUBSCRIBE, NOTIFY Supported: replaces Via: SIP/2.0/TCP 10.103.4.46;branch=z9hG4bK-sipXecs-31c6694e85f0cc87509492b2901411cc3ca0 Via: SIP/2.0/TCP 10.103.4.46;branch=z9hG4bK-sipXecs-31c34b9baa0e8096d2b2ac87546126cf5fe6~035dc386894103794b1cf6ab34cc9213 Via: SIP/2.0/TCP 10.103.4.46;branch=z9hG4bK-sipXecs-31be5b36f3fef612f87f2837b5e6593ca55f~593dbe2f4150ade23bda392ddcb8e4db Via: SIP/2.0/UDP 10.103.4.46:5150;branch=z9hG4bK-sipXecs-68aa8db1dab2755236e6f169d3ab9bdb028b Expires: 60 v=0 o=sipXecs 5 189 IN IP4 10.103.4.46 s=phone-call c=IN IP4 10.103.4.46 t=0 0 m=audio 10034 RTP/AVP 0 8 96 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
