========================================================================================== Incoming Call from Trunk ========================================================================================== <--- SIP read from UDP:94.75.247.45:5060 ---> INVITE sip:s@198.23.139.21:5080 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0 Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0 Max-Forwards: 12 From: ;tag=6pg6jZ1Q6Ky7c To: Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58 CSeq: 44635656 INVITE Contact: User-Agent: PAETEC Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY Supported: precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 253 X-SIPE: 62997 v=0 o=FreeSWITCH 1369924508 1369924509 IN IP4 66.217.168.172 s=FreeSWITCH c=IN IP4 66.217.168.172 t=0 0 m=audio 18932 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (18 headers 11 lines) --- Sending to 94.75.247.45:5060 (no NAT) Sending to 94.75.247.45:5060 (no NAT) Using INVITE request as basis request - 0c092aee-4405-1231-64a0-0030489f3d58 Found peer 'LPhone' for '4940306988122' from 94.75.247.45:5060 == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 13 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|silk8|silk12|silk16|silk24), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 66.217.168.172:18932 Peer doesn't provide video Looking for s in from-pstn-toheader (domain 198.23.139.21) list_route: route/path hop: <--- Transmitting (no NAT) to 94.75.247.45:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0;received=94.75.247.45 Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0 Record-Route: From: ;tag=6pg6jZ1Q6Ky7c To: Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58 CSeq: 44635656 INVITE Server: FPBX-2.11.0(1.8) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 198.23.139.21:5060: INVITE sip:1001@198.23.139.21:5060 SIP/2.0 Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK4f7baafe Max-Forwards: 70 From: "4940306988122" ;tag=as373889aa To: Contact: Call-ID: 10a421e0545886cc09b6af4a00c599d9@198.23.139.21:5080 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(1.8) Date: Thu, 30 May 2013 19:50:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 688 v=0 o=root 1520806685 1520806685 IN IP4 198.23.139.21 s=Asterisk PBX SVN-trunk-r389770 c=IN IP4 198.23.139.21 t=0 0 m=audio 11966 RTP/AVP 0 108 107 9 8 96 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:108 SILK/24000 a=fmtp:108 usedtx=0 a=fmtp:108 useinbandfec=1 a=rtpmap:107 SILK/16000 a=fmtp:107 maxaveragebitrate=30000 a=fmtp:107 usedtx=0 a=fmtp:107 useinbandfec=1 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 SILK/8000 a=fmtp:96 maxaveragebitrate=15000 a=fmtp:96 usedtx=0 a=fmtp:96 useinbandfec=1 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called SIP/1001 <--- Transmitting (no NAT) to 94.75.247.45:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0;received=94.75.247.45 Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0 Record-Route: From: ;tag=6pg6jZ1Q6Ky7c To: ;tag=as199823a4 Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58 CSeq: 44635656 INVITE Server: FPBX-2.11.0(1.8) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> <--- SIP read from UDP:198.23.139.21:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK4f7baafe From: "4940306988122" ;tag=as373889aa To: Call-ID: 10a421e0545886cc09b6af4a00c599d9@198.23.139.21:5080 CSeq: 102 INVITE Server: kamailio (4.0.1 (x86_64/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:198.23.139.21:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK4f7baafe Record-Route: Record-Route: Contact: To: ;tag=d0272262 From: "4940306988122";tag=as373889aa Call-ID: 10a421e0545886cc09b6af4a00c599d9@198.23.139.21:5080 CSeq: 102 INVITE User-Agent: X-Lite release 4.5.2 stamp 70142 Allow-Events: hold, talk Content-Length: 0 <-------------> --- (12 headers 0 lines) --- list_route: route/path hop: list_route: route/path hop: -- SIP/1001-00000007 is ringing <--- Transmitting (no NAT) to 94.75.247.45:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0;received=94.75.247.45 Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0 Record-Route: From: ;tag=6pg6jZ1Q6Ky7c To: ;tag=as199823a4 Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58 CSeq: 44635656 INVITE Server: FPBX-2.11.0(1.8) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Michael Leuker" ;party=called;privacy=off;screen=no Content-Length: 0 <------------> <--- SIP read from UDP:198.23.139.21:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK4f7baafe Record-Route: Record-Route: Contact: From: "4940306988122" ;tag=as373889aa Call-ID: 10a421e0545886cc09b6af4a00c599d9@198.23.139.21:5080 CSeq: 102 INVITE To: ;tag=A88EC20F1B7AAFA8A5F615621E3AD38D Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY Supported: replaces, path Content-Length: 0 <-------------> --- (12 headers 0 lines) --- list_route: route/path hop: list_route: route/path hop: -- SIP/1001-00000007 is ringing <--- SIP read from UDP:198.23.139.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK4f7baafe Record-Route: Record-Route: Contact: To: ;tag=d0272262 From: "4940306988122";tag=as373889aa Call-ID: 10a421e0545886cc09b6af4a00c599d9@198.23.139.21:5080 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, eventlist User-Agent: X-Lite release 4.5.2 stamp 70142 Content-Length: 246 v=0 o=- 13014417040298814 3 IN IP4 192.168.178.33 s=X-Lite 4 release 4.5.2 stamp 70142 c=IN IP4 188.105.112.187 t=0 0 m=audio 59026 RTP/AVP 0 9 8 97 101 a=rtpmap:97 ILBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (14 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 101 Found audio description format ILBC for ID 97 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|ilbc|g722|silk8|silk16|silk24), peer - audio=(ulaw|alaw|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|ilbc|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 188.105.112.187:59026 list_route: route/path hop: list_route: route/path hop: set_destination: Parsing for address/port to send to set_destination: set destination to 198.23.139.21:5060 Transmitting (no NAT) to 198.23.139.21:5060: ACK sip:1001@188.105.112.187:59986;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 198.23.139.21:5080;branch=z9hG4bK248efa9e Route: , Max-Forwards: 70 From: "4940306988122" ;tag=as373889aa To: ;tag=d0272262 Contact: Call-ID: 10a421e0545886cc09b6af4a00c599d9@198.23.139.21:5080 CSeq: 102 ACK User-Agent: FPBX-2.11.0(1.8) Content-Length: 0 --- -- SIP/1001-00000007 answered SIP/LPhone-00000006 Audio is at 11104 Adding codec 100003 (ulaw) to SDP Adding codec 100008 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 94.75.247.45:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0;received=94.75.247.45 Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0 Record-Route: From: ;tag=6pg6jZ1Q6Ky7c To: ;tag=as199823a4 Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58 CSeq: 44635656 INVITE Server: FPBX-2.11.0(1.8) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Michael Leuker" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 320 v=0 o=root 833496020 833496020 IN IP4 198.23.139.21 s=Asterisk PBX SVN-trunk-r389770 c=IN IP4 198.23.139.21 t=0 0 m=audio 11104 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> > 0x7f7ce419e9e0 -- Probation passed - setting RTP source address to 188.105.112.187:59026 Retransmitting #1 (no NAT) to 94.75.247.45:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.0;received=94.75.247.45 Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.0 Record-Route: From: ;tag=6pg6jZ1Q6Ky7c To: ;tag=as199823a4 Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58 CSeq: 44635656 INVITE Server: FPBX-2.11.0(1.8) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Michael Leuker" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 320 v=0 o=root 833496020 833496020 IN IP4 198.23.139.21 s=Asterisk PBX SVN-trunk-r389770 c=IN IP4 198.23.139.21 t=0 0 m=audio 11104 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- > 0x7f7ca4026610 -- Probation passed - setting RTP source address to 66.217.168.172:18932 <--- SIP read from UDP:94.75.247.45:5060 ---> ACK sip:s@198.23.139.21:5080 SIP/2.0 Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.2 Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.2 Max-Forwards: 12 From: ;tag=6pg6jZ1Q6Ky7c To: ;tag=as199823a4 Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58 CSeq: 44635656 ACK Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:94.75.247.45:5060 ---> ACK sip:s@198.23.139.21:5080 SIP/2.0 Via: SIP/2.0/UDP 94.75.247.45;branch=z9hG4bKb56d.8cc10bb5.2 Via: SIP/2.0/UDP 95.211.119.240;rport=5060;branch=z9hG4bKb56d.08ec6a26.2 Max-Forwards: 12 From: ;tag=6pg6jZ1Q6Ky7c To: ;tag=as199823a4 Call-ID: 0c092aee-4405-1231-64a0-0030489f3d58 CSeq: 44635656 ACK Contact: Content-Length: 0