myhostname*CLI> [May 18 23:32:16] <--- SIP read from UDP:192.168.10.187:44142 ---> INVITE sip:2001@test.local SIP/2.0 Call-ID: 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 CSeq: 2969 INVITE From: "2005" ;tag=2459195543 To: Via: SIP/2.0/UDP 192.168.10.187:44142;branch=z9hG4bKe1dd8d3efa42546a14508930a118e70034;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: "2005" Content-Type: application/sdp Content-Length: 300 v=0 o=- 1305754335548 1305754335550 IN IP4 192.168.10.187 s=- c=IN IP4 192.168.10.187 t=0 0 m=audio 7796 RTP/AVP 96 97 3 0 8 127 a=rtpmap:96 GSM-EFR/8000 a=rtpmap:97 AMR/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> [May 18 23:32:16] --- (11 headers 13 lines) --- [May 18 23:32:16] == Using UDPTL CoS mark 5 [May 18 23:32:16] Sending to 192.168.10.187:44142 (no NAT) [May 18 23:32:16] Using INVITE request as basis request - 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 [May 18 23:32:16] Found peer '2005' for '2005' from 192.168.10.187:44142 [May 18 23:32:16] <--- Reliably Transmitting (NAT) to 192.168.10.187:44142 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.187:44142;branch=z9hG4bKe1dd8d3efa42546a14508930a118e70034;received=192.168.10.187;rport=44142 From: "2005" ;tag=2459195543 To: ;tag=as3b32f291 Call-ID: 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 CSeq: 2969 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="myhostname", nonce="2a544f0f" Content-Length: 0 <------------> [May 18 23:32:16] Scheduling destruction of SIP dialog '25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187' in 32000 ms (Method: INVITE) [May 18 23:32:17] <--- SIP read from UDP:192.168.10.187:44142 ---> ACK sip:2001@test.local SIP/2.0 Call-ID: 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 Max-Forwards: 70 From: "2005" ;tag=2459195543 To: ;tag=as3b32f291 Via: SIP/2.0/UDP 192.168.10.187:44142;branch=z9hG4bKe1dd8d3efa42546a14508930a118e70034;rport CSeq: 2969 ACK Content-Length: 0 <-------------> [May 18 23:32:17] --- (8 headers 0 lines) --- [May 18 23:32:17] <--- SIP read from UDP:192.168.10.187:44142 ---> INVITE sip:2001@test.local:5060 SIP/2.0 Call-ID: 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 CSeq: 2970 INVITE From: "2005" ;tag=2459195543 To: Via: SIP/2.0/UDP 192.168.10.187:44142;branch=z9hG4bKea4e9b6946fc10eaebd189b142eeca7a34;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: "2005" Content-Type: application/sdp Authorization: Digest username="2005",realm="myhostname",nonce="2a544f0f",uri="sip:2001@test.local:5060",response="e0588357cbf6fc0bd289fecde5b5090e",algorithm=MD5 Content-Length: 300 v=0 o=- 1305754335548 1305754335550 IN IP4 192.168.10.187 s=- c=IN IP4 192.168.10.187 t=0 0 m=audio 7796 RTP/AVP 96 97 3 0 8 127 a=rtpmap:96 GSM-EFR/8000 a=rtpmap:97 AMR/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> [May 18 23:32:17] --- (12 headers 13 lines) --- [May 18 23:32:17] == Using UDPTL CoS mark 5 [May 18 23:32:17] Sending to 192.168.10.187:44142 (no NAT) [May 18 23:32:17] Using INVITE request as basis request - 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 [May 18 23:32:17] Found peer '2005' for '2005' from 192.168.10.187:44142 [May 18 23:32:17] <--- Reliably Transmitting (NAT) to 192.168.10.187:44142 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.187:44142;branch=z9hG4bKea4e9b6946fc10eaebd189b142eeca7a34;received=192.168.10.187;rport=44142 From: "2005" ;tag=2459195543 To: ;tag=as2665ed9c Call-ID: 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 CSeq: 2970 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="myhostname", nonce="xxxxx" Content-Length: 0 <------------> [May 18 23:32:17] Scheduling destruction of SIP dialog '25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187' in 32000 ms (Method: INVITE) [May 18 23:32:17] <--- SIP read from UDP:192.168.10.187:44142 ---> ACK sip:2001@test.local:5060 SIP/2.0 Call-ID: 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 Max-Forwards: 70 From: "2005" ;tag=2459195543 To: ;tag=as2665ed9c Via: SIP/2.0/UDP 192.168.10.187:44142;branch=z9hG4bKea4e9b6946fc10eaebd189b142eeca7a34;rport CSeq: 2970 ACK Content-Length: 0 <-------------> [May 18 23:32:17] --- (8 headers 0 lines) --- [May 18 23:32:17] <--- SIP read from UDP:192.168.10.187:44142 ---> INVITE sip:2001@test.local:5060 SIP/2.0 Call-ID: 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 CSeq: 2971 INVITE From: "2005" ;tag=2459195543 To: Via: SIP/2.0/UDP 192.168.10.187:44142;branch=z9hG4bK95098c98e2a7f064a1438d784561680234;rport Max-Forwards: 70 User-Agent: SIPAUA/0.1.001 Contact: "2005" Content-Type: application/sdp Authorization: Digest username="2005",realm="myhostname",nonce="xxxxx",uri="sip:2001@test.local:5060",response="xxxxxxx",algorithm=MD5 Content-Length: 300 v=0 o=- 1305754335548 1305754335550 IN IP4 192.168.10.187 s=- c=IN IP4 192.168.10.187 t=0 0 m=audio 7796 RTP/AVP 96 97 3 0 8 127 a=rtpmap:96 GSM-EFR/8000 a=rtpmap:97 AMR/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> [May 18 23:32:17] --- (12 headers 13 lines) --- [May 18 23:32:17] Sending to 192.168.10.187:44142 (NAT) [May 18 23:32:17] Using INVITE request as basis request - 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 [May 18 23:32:17] Found peer '2005' for '2005' from 192.168.10.187:44142 [May 18 23:32:17] == Using SIP RTP CoS mark 5 [May 18 23:32:17] Found RTP audio format 96 [May 18 23:32:17] Found RTP audio format 97 [May 18 23:32:17] Found RTP audio format 3 [May 18 23:32:17] Found RTP audio format 0 [May 18 23:32:17] Found RTP audio format 8 [May 18 23:32:17] Found RTP audio format 127 [May 18 23:32:17] Found audio description format GSM-EFR for ID 96 [May 18 23:32:17] Found audio description format AMR for ID 97 [May 18 23:32:17] Found audio description format GSM for ID 3 [May 18 23:32:17] Found audio description format PCMU for ID 0 [May 18 23:32:17] Found audio description format PCMA for ID 8 [May 18 23:32:17] Found audio description format telephone-event for ID 127 [May 18 23:32:17] Capabilities: us - 0x20030e (gsm|ulaw|alaw|g729|speex|h264), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [May 18 23:32:17] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 18 23:32:17] Peer audio RTP is at port 192.168.10.187:7796 [May 18 23:32:17] Peer doesn't provide video [May 18 23:32:17] Looking for 2001 in internal (domain test.local:5060) [May 18 23:32:17] list_route: hop: [May 18 23:32:17] <--- Transmitting (NAT) to 192.168.10.187:44142 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.187:44142;branch=z9hG4bK95098c98e2a7f064a1438d784561680234;received=192.168.10.187;rport=44142 From: "2005" ;tag=2459195543 To: Call-ID: 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 CSeq: 2971 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [May 18 23:32:17] -- Executing [2001@internal:1] NoOp("SIP/2005-0000000a", "") in new stack [May 18 23:32:17] -- Executing [2001@internal:2] Dial("SIP/2005-0000000a", "SIP/2001") in new stack [May 18 23:32:17] == Using UDPTL CoS mark 5 [May 18 23:32:17] == Using SIP RTP CoS mark 5 [May 18 23:32:17] -- Called 2001 [May 18 23:32:17] -- SIP/2001-0000000b is ringing [May 18 23:32:17] <--- Transmitting (NAT) to 192.168.10.187:44142 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.187:44142;branch=z9hG4bK95098c98e2a7f064a1438d784561680234;received=192.168.10.187;rport=44142 From: "2005" ;tag=2459195543 To: ;tag=as2cc6da29 Call-ID: 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 CSeq: 2971 INVITE Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> myhostname*CLI> myhostname*CLI> myhostname*CLI> myhostname*CLI> [May 18 23:32:21] <--- SIP read from UDP:192.168.10.187:44142 ---> CANCEL sip:2001@test.local:5060 SIP/2.0 Call-ID: 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 To: CSeq: 2971 CANCEL From: "2005" ;tag=2459195543 Via: SIP/2.0/UDP 192.168.10.187:44142;branch=z9hG4bK95098c98e2a7f064a1438d784561680234;rport Max-Forwards: 70 Content-Length: 0 <-------------> [May 18 23:32:21] --- (8 headers 0 lines) --- [May 18 23:32:21] <--- Transmitting (NAT) to 192.168.10.187:44142 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.10.187:44142;branch=z9hG4bK95098c98e2a7f064a1438d784561680234;received=192.168.10.187;rport=44142 From: "2005" ;tag=2459195543 To: ;tag=as3b32f291 Call-ID: 25f1c19c6e0a45b5713281c0e34256c6@192.168.10.187 CSeq: 2971 CANCEL Server: Asterisk PBX 1.8.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> myhostname*CLI>