Reliably Transmitting (NAT) tomy.provider.com:5060: INVITE sip:To_num@my.provider.com SIP/2.0 Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK541a7830;rport Max-Forwards: 70 From: "203" ;tag=as17dca408 To: Contact: Call-ID: 4458465f68af28ca0ccd8a073120c228@my.provider.com CSeq: 102 INVITE User-Agent: Asterisk PBX 11.11.0 Date: Sun, 07 Sep 2014 00:15:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 283 v=0 o=root 1016367194 1016367194 IN IP4 my.server.com s=Asterisk PBX 11.11.0 c=IN IP4 my.server.com t=0 0 m=audio 18258 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <--- SIP read from UDP:my.provider.com:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK541a7830;rport=5068 From: "203" ;tag=as17dca408 To: ;tag=7aca57471283d03f3321fdece75f8666.0554 Call-ID: 4458465f68af28ca0ccd8a073120c228@my.provider.com CSeq: 102 INVITE Proxy-Authenticate: Digest realm="my.provider.com", nonce="VAukwFQLo5TCvnt9GWaCSWg8ULPgIzU9", qop="auth" Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (NAT) tomy.provider.com:5060: ACK sip:To_num@my.provider.com SIP/2.0 Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK541a7830;rport Max-Forwards: 70 From: "203" ;tag=as17dca408 To: ;tag=7aca57471283d03f3321fdece75f8666.0554 Contact: Call-ID: 4458465f68af28ca0ccd8a073120c228@my.provider.com CSeq: 102 ACK User-Agent: Asterisk PBX 11.11.0 Content-Length: 0 Reliably Transmitting (NAT) tomy.provider.com:5060: INVITE sip:To_num@my.provider.com SIP/2.0 Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK3a86505a;rport Max-Forwards: 70 From: "203" ;tag=as17dca408 To: Contact: Call-ID: 4458465f68af28ca0ccd8a073120c228@my.provider.com CSeq: 103 INVITE User-Agent: Asterisk PBX 11.11.0 Proxy-Authorization: Digest username="From_num", realm="my.provider.com", algorithm=MD5, uri="sip:To_num@my.provider.com", nonce="VAukwFQLo5TCvnt9GWaCSWg8ULPgIzU9", response="c21f09cb598343b7c4638de6f0aa5678", qop=auth, cnonce="7fd459c6", nc=00000001 Date: Sun, 07 Sep 2014 00:15:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 283 v=0 o=root 1016367194 1016367195 IN IP4 my.server.com s=Asterisk PBX 11.11.0 c=IN IP4 my.server.com t=0 0 m=audio 18258 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:my.provider.com:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK3a86505a;rport=5068 From: "203" ;tag=as17dca408 To: Call-ID: 4458465f68af28ca0ccd8a073120c228@my.provider.com CSeq: 103 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 <--- SIP read from UDP:my.provider.com:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK3a86505a;rport=5068 Record-Route: From: "203" ;tag=as17dca408 To: ;tag=ZZ7H26FHvvQDe Call-ID: 4458465f68af28ca0ccd8a073120c228@my.provider.com CSeq: 103 INVITE Contact: User-Agent: Plivo Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 Remote-Party-ID: "To_num" ;party=calling;privacy=off;screen=no <--- SIP read from UDP:my.provider.com:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK3a86505a;rport=5068 Record-Route: From: "203" ;tag=as17dca408 To: ;tag=ZZ7H26FHvvQDe Call-ID: 4458465f68af28ca0ccd8a073120c228@my.provider.com CSeq: 103 INVITE Contact: User-Agent: Plivo Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 253 Remote-Party-ID: "To_num" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1410020988 1410020989 IN IP4 212.100.254.141 s=FreeSWITCH c=IN IP4 212.100.254.141 t=0 0 m=audio 27930 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 <-------------> <--- SIP read from UDP:my.provider.com:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK3a86505a;rport=5068 Record-Route: From: "203" ;tag=as17dca408 To: ;tag=ZZ7H26FHvvQDe Call-ID: 4458465f68af28ca0ccd8a073120c228@my.provider.com CSeq: 103 INVITE Contact: User-Agent: Plivo Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 253 X-PlivoOutboundGateway: sip:55740To_num@62.93.147.149 X-PlivoOutboundCarrierID: 23705946361020 X-PlivoCarrierRate: 0.20180 X-PlivoCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1410020988 1410020989 IN IP4 212.100.254.141 s=FreeSWITCH c=IN IP4 212.100.254.141 t=0 0 m=audio 27930 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ACK sip:To_num@212.100.254.141:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.server.com:5068;branch=z9hG4bK09fc11af;rport Route: Max-Forwards: 70 From: "203" ;tag=as17dca408 To: ;tag=ZZ7H26FHvvQDe Contact: Call-ID: 4458465f68af28ca0ccd8a073120c228@my.provider.com CSeq: 103 ACK User-Agent: Asterisk PBX 11.11.0 Content-Length: 0 <--- SIP read from UDP:my.provider.com:5060 ---> BYE sip:From_num@my.server.com:5068 SIP/2.0 Record-Route: Via: SIP/2.0/UDPmy.provider.com:5060;branch=z9hG4bK43a4.4e3a49dd8c75379912ac6ff1a9904244.0 Via: SIP/2.0/UDP 212.100.254.141;rport=5060;branch=z9hG4bKaH07HF0SavaDN Max-Forwards: 16 From: ;tag=ZZ7H26FHvvQDe To: "203" ;tag=as17dca408 Call-ID: 4458465f68af28ca0ccd8a073120c228@my.provider.com CSeq: 64688402 BYE Contact: User-Agent: Plivo Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=16 Content-Length: 0 <--- Transmitting (NAT) tomy.provider.com:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDPmy.provider.com:5060;branch=z9hG4bK43a4.4e3a49dd8c75379912ac6ff1a9904244.0;received=my.provider.com;rport=5060 Via: SIP/2.0/UDP 212.100.254.141;rport=5060;branch=z9hG4bKaH07HF0SavaDN Record-Route: From: ;tag=ZZ7H26FHvvQDe To: "203" ;tag=as17dca408 Call-ID: 4458465f68af28ca0ccd8a073120c228@my.provider.com CSeq: 64688402 BYE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0