<-------------> [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: --- (19 headers 15 lines) --- [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Sending to 172.17.0.143:5060 (no NAT) [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Using INVITE request as basis request - 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found peer '7662' for '7662' from 172.17.0.143:49160 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: <--- Reliably Transmitting (no NAT) to 172.17.0.143:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK751df164;received=172.17.0.143 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: ;tag=as19efe676 Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 CSeq: 101 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="ATS", nonce="049f13bf" Content-Length: 0 <------------> [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Scheduling destruction of SIP dialog '002584a0-c4810025-b38f72d3-5d660918@172.17.0.143' in 6400 ms (Method: INVITE) [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: <--- SIP read from UDP:172.17.0.143:50587 ---> ACK sip:7664@172.17.0.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK751df164 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: ;tag=as19efe676 Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 Date: Thu, 27 Jan 2011 13:16:20 GMT CSeq: 101 ACK Content-Length: 0 <-------------> [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: --- (8 headers 0 lines) --- [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: <--- SIP read from UDP:172.17.0.143:49160 ---> INVITE sip:7664@172.17.0.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 Max-Forwards: 70 Date: Thu, 27 Jan 2011 13:16:20 GMT CSeq: 102 INVITE User-Agent: Cisco-CP7911G/8.3.0 Contact: Authorization: Digest username="7662",realm="ATS",uri="sip:7664@172.17.0.10;user=phone",response="b36daa1cc32163d595a92cbd988137bd",nonce="049f13bf",algorithm=MD5 Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "RJTN D" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Allow-Events: kpml,dialog Content-Length: 324 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 27178 0 IN IP4 172.17.0.143 s=SIP Call t=0 0 m=audio 16472 RTP/AVP 0 8 18 116 101 c=IN IP4 172.17.0.143 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: --- (20 headers 15 lines) --- [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Sending to 172.17.0.143:5060 (no NAT) [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Using INVITE request as basis request - 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found peer '7662' for '7662' from 172.17.0.143:49160 [Jan 27 17:16:25] VERBOSE[18496] netsock2.c: == Using SIP RTP TOS bits 184 [Jan 27 17:16:25] VERBOSE[18496] netsock2.c: == Using SIP RTP CoS mark 5 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found RTP audio format 0 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found RTP audio format 8 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found RTP audio format 18 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found RTP audio format 116 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found RTP audio format 101 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found audio description format G729 for ID 18 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found audio description format iLBC for ID 116 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Capabilities: us - 0x180108 (alaw|g729|h263|h263p), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729) [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Peer audio RTP is at port 172.17.0.143:16472 [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Peer doesn't provide video [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: Looking for 7664 in international (domain 172.17.0.10) [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: list_route: hop: [Jan 27 17:16:25] VERBOSE[18496] chan_sip.c: <--- Transmitting (no NAT) to 172.17.0.143:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f;received=172.17.0.143 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 27 17:16:25] VERBOSE[18580] pbx.c: -- Executing [7664@international:1] GotoIfTime("SIP/7662-0000001f", "10:45-12:30,Fri,1-31,Jan-Dec?internal,667,1") in new stack [Jan 27 17:16:25] VERBOSE[18580] pbx.c: -- Executing [7664@international:2] Dial("SIP/7662-0000001f", "SIP/7664,25,tTo") in new stack [Jan 27 17:16:25] VERBOSE[18580] netsock2.c: == Using SIP RTP TOS bits 184 [Jan 27 17:16:25] VERBOSE[18580] netsock2.c: == Using SIP RTP CoS mark 5 [Jan 27 17:16:25] VERBOSE[18580] app_dial.c: -- Called 7664 [Jan 27 17:16:25] VERBOSE[18580] app_dial.c: -- SIP/7664-00000020 is ringing [Jan 27 17:16:25] VERBOSE[18580] chan_sip.c: <--- Transmitting (no NAT) to 172.17.0.143:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f;received=172.17.0.143 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: ;tag=as2ddab26c Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Jaskaran N" ;party=called;privacy=off;screen=yes Content-Length: 0 <------------> [Jan 27 17:16:28] VERBOSE[18574] pbx.c: -- Executing [h@local:1] SoftHangup("SIP/7724-0000001a", "SIP/7724-0000001a") in new stack [Jan 27 17:16:28] VERBOSE[18574] pbx.c: == Spawn extension (local, 07150776, 8) exited non-zero on 'SIP/7724-0000001a' [Jan 27 17:16:36] VERBOSE[18496] chan_sip.c: <--- SIP read from UDP:172.17.0.143:49160 ---> CANCEL sip:7664@172.17.0.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 Max-Forwards: 70 Date: Thu, 27 Jan 2011 13:16:32 GMT CSeq: 102 CANCEL User-Agent: Cisco-CP7911G/8.3.0 Content-Length: 0 Authorization: Digest username="7662",realm="ATS",uri="sip:7664@172.17.0.10;user=phone",response="dfecfe2049cb53748dda7c24a4781fcd",nonce="049f13bf",algorithm=MD5 <-------------> [Jan 27 17:16:36] VERBOSE[18496] chan_sip.c: --- (11 headers 0 lines) --- [Jan 27 17:16:36] VERBOSE[18496] chan_sip.c: Sending to 172.17.0.143:5060 (no NAT) [Jan 27 17:16:36] VERBOSE[18496] chan_sip.c: <--- Reliably Transmitting (no NAT) to 172.17.0.143:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f;received=172.17.0.143 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: ;tag=as2ddab26c Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 27 17:16:36] VERBOSE[18496] chan_sip.c: <--- Transmitting (no NAT) to 172.17.0.143:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f;received=172.17.0.143 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: ;tag=as2ddab26c Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 CSeq: 102 CANCEL Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 27 17:16:36] VERBOSE[18580] pbx.c: == Spawn extension (international, 7664, 2) exited non-zero on 'SIP/7662-0000001f' [Jan 27 17:16:36] VERBOSE[18580] pbx.c: -- Executing [h@international:1] SoftHangup("SIP/7662-0000001f", "SIP/7662-0000001f") in new stack [Jan 27 17:16:36] VERBOSE[18496] chan_sip.c: <--- SIP read from UDP:172.17.0.143:49160 ---> ACK sip:7664@172.17.0.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK8e16bfed From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 Max-Forwards: 70 Date: Thu, 27 Jan 2011 13:16:32 GMT CSeq: 102 ACK User-Agent: Cisco-CP7911G/8.3.0 Authorization: Digest username="7662",realm="ATS",uri="sip:7664@172.17.0.10;user=phone",response="dfecfe2049cb53748dda7c24a4781fcd",nonce="049f13bf",algorithm=MD5 Remote-Party-ID: "RJTN D" ;party=calling;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 <-------------> [Jan 27 17:16:36] VERBOSE[18496] chan_sip.c: --- (12 headers 0 lines) --- [Jan 27 17:16:36] VERBOSE[18496] chan_sip.c: Retransmitting #1 (no NAT) to 172.17.0.143:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f;received=172.17.0.143 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: ;tag=as2ddab26c Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jan 27 17:16:37] VERBOSE[18496] chan_sip.c: Retransmitting #2 (no NAT) to 172.17.0.143:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f;received=172.17.0.143 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: ;tag=as2ddab26c Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jan 27 17:16:37] VERBOSE[18496] chan_sip.c: Retransmitting #3 (no NAT) to 172.17.0.143:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f;received=172.17.0.143 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: ;tag=as2ddab26c Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jan 27 17:16:38] VERBOSE[18496] chan_sip.c: Retransmitting #4 (no NAT) to 172.17.0.143:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f;received=172.17.0.143 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: ;tag=as2ddab26c Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jan 27 17:16:38] VERBOSE[18496] chan_sip.c: Reliably Transmitting (no NAT) to 172.17.0.143:5060: OPTIONS sip:7662@172.17.0.143:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.0.10:5060;branch=z9hG4bK56bd04f9 Max-Forwards: 70 From: "asterisk" ;tag=as40a99a2d To: Contact: Call-ID: 078b47024d7254653a294c604a1d341a@172.17.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.3 Date: Thu, 27 Jan 2011 13:16:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jan 27 17:16:38] VERBOSE[18496] chan_sip.c: <--- SIP read from UDP:172.17.0.143:50588 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.0.10:5060;branch=z9hG4bK56bd04f9 From: "asterisk" ;tag=as40a99a2d To: ;tag=002584a0c48105e32571d197-101dfce1 Call-ID: 078b47024d7254653a294c604a1d341a@172.17.0.10:5060 Date: Thu, 27 Jan 2011 13:16:33 GMT CSeq: 102 OPTIONS Server: Cisco-CP7911G/8.3.0 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Allow-Events: kpml,dialog,refer Accept: application/sdp,multipart/mixed,multipart/alternative Accept-Encoding: identity Accept-Language: en Supported: replaces,join,norefersub Content-Length: 285 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 26523 0 IN IP4 172.17.0.143 s=SIP Call t=0 0 m=audio 0 RTP/AVP 0 8 18 116 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Jan 27 17:16:38] VERBOSE[18496] chan_sip.c: --- (17 headers 13 lines) --- [Jan 27 17:16:38] VERBOSE[18496] chan_sip.c: Really destroying SIP dialog '078b47024d7254653a294c604a1d341a@172.17.0.10:5060' Method: OPTIONS [Jan 27 17:16:38] VERBOSE[18577] pbx.c: -- Executing [h@international:1] SoftHangup("SIP/7650-0000001d", "SIP/7650-0000001d") in new stack [Jan 27 17:16:38] VERBOSE[18577] pbx.c: == Spawn extension (international, 7634, 2) exited non-zero on 'SIP/7650-0000001d' [Jan 27 17:16:38] VERBOSE[18579] asterisk.c: -- Remote UNIX connection disconnected [Jan 27 17:16:39] VERBOSE[18496] chan_sip.c: Retransmitting #5 (no NAT) to 172.17.0.143:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f;received=172.17.0.143 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: ;tag=as2ddab26c Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jan 27 17:16:43] VERBOSE[18496] chan_sip.c: Retransmitting #6 (no NAT) to 172.17.0.143:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 172.17.0.143:5060;branch=z9hG4bK5cec426f;received=172.17.0.143 From: "RJTN D" ;tag=002584a0c48105e2a01782a1-c30a73b2 To: ;tag=as2ddab26c Call-ID: 002584a0-c4810025-b38f72d3-5d660918@172.17.0.143 CSeq: 102 INVITE Server: Asterisk PBX 1.8.2.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jan 27 17:16:43] VERBOSE[18496] chan_sip.c: Really destroying SIP dialog '002584a0-c4810025-b38f72d3-5d660918@172.17.0.143' Method: CANCEL