**** Edited to anonymize domain URLs, phone numbers and userids ***** <--- SIP read from UDP:166.137.87.102:61089 ---> INVITE sip:*1@pbx.domain.org SIP/2.0 Via: SIP/2.0/UDP 10.10.242.5:2034;branch=z9hG4bKu8cr2TDbjzwV6T7Y;rport Contact: Max-Forwards: 70 From: "David" ;tag=0EA9FBFAD87083C1C7CFFCA6BB22C2F9 Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY Supported: replaces, path User-Agent: Acrobits Softphone/5.2 To: Content-Type: application/sdp Call-ID: D084B497A0A9430F7ABB0D3DA7A7D78AC60B20F9 CSeq: 1 INVITE Content-Length: 240 v=0 o=- 19412 50398 IN IP4 10.10.242.5 s=mkwhemp c=IN IP4 10.10.242.5 t=0 0 m=audio 19108 RTP/AVP 102 3 0 8 9 101 a=rtpmap:101 telephone-event/8000 a=rtpmap:102 ILBC/8000 a=fmtp:102 mode=30 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (13 headers 12 lines) --- Sending to 166.137.87.102:61089 (NAT) Using INVITE request as basis request - D084B497A0A9430F7ABB0D3DA7A7D78AC60B20F9 Found peer '104' for '104' from 166.137.87.102:61089 == Using SIP RTP CoS mark 5 Found RTP audio format 102 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Found audio description format ILBC for ID 102 Capabilities: us - 0x1506 (gsm|ulaw|g729|ilbc|g722), peer - audio=0x140e (gsm|ulaw|alaw|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1406 (gsm|ulaw|ilbc|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.10.242.5:19108 Looking for *1 in DialPlanSLA (domain pbx.domain.org) list_route: hop: <--- Transmitting (NAT) to 166.137.87.102:61089 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.242.5:2034;branch=z9hG4bKu8cr2TDbjzwV6T7Y;received=166.137.87.102;rport=61089 From: "David" ;tag=0EA9FBFAD87083C1C7CFFCA6BB22C2F9 To: Call-ID: D084B497A0A9430F7ABB0D3DA7A7D78AC60B20F9 CSeq: 1 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [*1@DialPlanSLA:1] SLAStation("SIP/104-0000004e", "SLAphone104_SLAtrunk1") in new stack -- Called s@OutboundSLA -- Executing [s@OutboundSLA:1] NoOp("Local/s@OutboundSLA-df6c;2", "SLA Outbound context") in new stack -- Executing [s@OutboundSLA:2] DISA("Local/s@OutboundSLA-df6c;2", "no-password,OutboundSLA") in new stack -- Local/s@OutboundSLA-df6c;1 answered -- Created MeetMe conference 1023 for conference 'SLA_SLAtrunk1' Audio is at 16394 Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x1000 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 166.137.87.102:61089 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.242.5:2034;branch=z9hG4bKu8cr2TDbjzwV6T7Y;received=166.137.87.102;rport=61089 From: "David" ;tag=0EA9FBFAD87083C1C7CFFCA6BB22C2F9 To: ;tag=as6420d410 Call-ID: D084B497A0A9430F7ABB0D3DA7A7D78AC60B20F9 CSeq: 1 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 334 v=0 o=root 2141099548 2141099548 IN IP4 24.128.119.26 s=Asterisk PBX 1.8.15.0 c=IN IP4 24.128.119.26 t=0 0 m=audio 16394 RTP/AVP 102 3 0 9 101 a=rtpmap:102 iLBC/8000 a=fmtp:102 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:166.137.87.102:61089 ---> ACK sip:*1@24.128.119.26:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.242.5:2034;branch=z9hG4bKhBQz9Nf9uEDHCZLb;rport Max-Forwards: 70 To: ;tag=as6420d410 From: "David" ;tag=0EA9FBFAD87083C1C7CFFCA6BB22C2F9 Call-ID: D084B497A0A9430F7ABB0D3DA7A7D78AC60B20F9 CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '51214729@192_168_17_141' Method: REGISTER Really destroying SIP dialog '1838558966@192_168_17_141' Method: REGISTER -- Executing [1914yyyxxxx@OutboundSLA:1] Macro("Local/s@OutboundSLA-df6c;2", "trunkdial-failover,SIP/vitel-outbound/1914yyyxxxx,SIP/voipcheap/001914yyyxxxx,vitel-outbound,voipcheap") in new stack -- Executing [s@macro-trunkdial-failover:1] NoOp("Local/s@OutboundSLA-df6c;2", "trunk dial by 104") in new stack -- Executing [s@macro-trunkdial-failover:2] ExecIf("Local/s@OutboundSLA-df6c;2", "0?Set(VOLUME(RX)=10)") in new stack -- Executing [s@macro-trunkdial-failover:3] ExecIf("Local/s@OutboundSLA-df6c;2", "0?Set(VOLUME(RX)=10)") in new stack -- Executing [s@macro-trunkdial-failover:4] GotoIf("Local/s@OutboundSLA-df6c;2", "0?1-fmsetcid,1") in new stack -- Executing [s@macro-trunkdial-failover:5] GotoIf("Local/s@OutboundSLA-df6c;2", "0?1-setgbobname,1") in new stack -- Executing [s@macro-trunkdial-failover:6] Set("Local/s@OutboundSLA-df6c;2", "CALLERID(num)=203yyyxxxx") in new stack -- Executing [s@macro-trunkdial-failover:7] GotoIf("Local/s@OutboundSLA-df6c;2", "1?1-dial,1") in new stack -- Goto (macro-trunkdial-failover,1-dial,1) -- Executing [1-dial@macro-trunkdial-failover:1] Dial("Local/s@OutboundSLA-df6c;2", "SIP/vitel-outbound/1914yyyxxxx,,rT") in new stack == Using SIP RTP CoS mark 5 Audio is at 16560 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 64.2.142.189:5060: INVITE sip:1914yyyxxxx@outbound.vitelity.net SIP/2.0 Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK425dbaee;rport Max-Forwards: 70 From: "Softphone" ;tag=as5536a4cd To: Contact: Call-ID: 18bf1819371d28b1659750b3021942e2@24.128.119.26:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.15.0 Date: Tue, 18 Sep 2012 19:32:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "Softphone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 286 v=0 o=root 1439404609 1439404609 IN IP4 24.128.119.26 s=Asterisk PBX 1.8.15.0 c=IN IP4 24.128.119.26 t=0 0 m=audio 16560 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=ptime:20 a=sendrecv --- -- Called SIP/vitel-outbound/1914yyyxxxx <--- SIP read from UDP:64.2.142.189:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK425dbaee;received=24.128.119.26;rport=5060 From: "Softphone" ;tag=as5536a4cd To: ;tag=as7a338b40 Call-ID: 18bf1819371d28b1659750b3021942e2@24.128.119.26:5060 CSeq: 102 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="464022aa" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (NAT) to 64.2.142.189:5060: ACK sip:1914yyyxxxx@outbound.vitelity.net SIP/2.0 Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK425dbaee;rport Max-Forwards: 70 From: "Softphone" ;tag=as5536a4cd To: ;tag=as7a338b40 Contact: Call-ID: 18bf1819371d28b1659750b3021942e2@24.128.119.26:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 --- Audio is at 16560 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 64.2.142.189:5060: INVITE sip:1914yyyxxxx@outbound.vitelity.net SIP/2.0 Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK7830a8bb;rport Max-Forwards: 70 From: "Softphone" ;tag=as5536a4cd To: Contact: Call-ID: 18bf1819371d28b1659750b3021942e2@24.128.119.26:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.15.0 Proxy-Authorization: Digest username="dk123", realm="asterisk", algorithm=MD5, uri="sip:1914yyyxxxx@outbound.vitelity.net", nonce="464022aa", response="2ea2f6381cfb62f43f540d415fb1bad4" Date: Tue, 18 Sep 2012 19:32:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "Softphone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 286 v=0 o=root 1439404609 1439404610 IN IP4 24.128.119.26 s=Asterisk PBX 1.8.15.0 c=IN IP4 24.128.119.26 t=0 0 m=audio 16560 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=ptime:20 a=sendrecv --- <--- SIP read from UDP:64.2.142.189:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK7830a8bb;received=24.128.119.26;rport=5060 From: "Softphone" ;tag=as5536a4cd To: Call-ID: 18bf1819371d28b1659750b3021942e2@24.128.119.26:5060 CSeq: 103 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:64.2.142.189:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK7830a8bb;received=24.128.119.26;rport=5060 From: "Softphone" ;tag=as5536a4cd To: ;tag=as13109d91 Call-ID: 18bf1819371d28b1659750b3021942e2@24.128.119.26:5060 CSeq: 103 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- list_route: hop: -- SIP/vitel-outbound-0000004f is ringing Really destroying SIP dialog '47a723b6639149626f3ec3df6360f6e5@192.168.17.1' Method: REGISTER <--- SIP read from UDP:64.2.142.189:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK7830a8bb;received=24.128.119.26;rport=5060 From: "Softphone" ;tag=as5536a4cd To: ;tag=as13109d91 Call-ID: 18bf1819371d28b1659750b3021942e2@24.128.119.26:5060 CSeq: 103 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 16008 16008 IN IP4 64.2.142.189 s=session c=IN IP4 64.2.142.189 t=0 0 m=audio 18392 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 <-------------> --- (12 headers 14 lines) --- Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 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 - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 64.2.142.189:18392 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 64.2.142.189:5060 Transmitting (NAT) to 64.2.142.189:5060: ACK sip:1914yyyxxxx@64.2.142.189 SIP/2.0 Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK270f3b90;rport Max-Forwards: 70 From: "Softphone" ;tag=as5536a4cd To: ;tag=as13109d91 Contact: Call-ID: 18bf1819371d28b1659750b3021942e2@24.128.119.26:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 --- -- SIP/vitel-outbound-0000004f answered Local/s@OutboundSLA-df6c;2 -- Executing [h@OutboundSLA:1] Hangup("Local/s@OutboundSLA-df6c;2", "") in new stack == Spawn extension (OutboundSLA, h, 1) exited non-zero on 'Local/s@OutboundSLA-df6c;2' == Spawn extension (macro-trunkdial-failover, 1-dial, 1) exited non-zero on 'Local/s@OutboundSLA-df6c;2' in macro 'trunkdial-failover' == Spawn extension (OutboundSLA, 1914yyyxxxx, 1) exited non-zero on 'Local/s@OutboundSLA-df6c;2'