Script started on ven 05 aoû 2005 15:17:46 CEST itbx:/usr/src# asterisk -rvvvvdddd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf Asterisk 1.0.9-0.2.0-itbx-8k, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk 1.0.9-0.2.0-itbx-8k currently running on itbx (pid = 24069) itbx*CLI> Verbosity is at least 4 Core debug is at least 4 itbx*CLI> sip debug peer 6001 itbx*CLI> SIP Debugging Enabled for IP: 172.17.100.248:5060 itbx*CLI> sip debug peer 60015 itbx*CLI> SIP Debugging Enabled for IP: 172.17.100.222:5080 itbx*CLI> itbx*CLI> -- Executing SetVar("SIP/6005-5184", "CALLEDID=6001") in new stack -- Executing GotoIf("SIP/6005-5184", "0?3:7") in new stack -- Goto (from-ip-phones,6001,7) -- Executing Macro("SIP/6005-5184", "dialuser|6001|30|rF") in new stack -- Executing DBget("SIP/6005-5184", "temp=FM/6001") in new stack -- DBget: varname=temp, family=FM, key=6001 -- DBget: set variable temp to 6001 -- Executing GotoIf("SIP/6005-5184", "1?102:3") in new stack -- Goto (macro-dialuser,s,102) itbx*CLI> -- Executing Dial("SIP/6005-5184", "SIP/6001|30|rF") in new stack -- Called 6001 itbx*CLI> -- SIP/6001-9ef2 is ringing itbx*CLI> -- SIP/6001-9ef2 answered SIP/6005-5184 -- Attempting native bridge of SIP/6005-5184 and SIP/6001-9ef2 itbx*CLI> Transmitting: ACK sip:6005@172.17.100.222:5083 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK60c7441e;rport From: ;tag=as64a2cdf9 To: "Damien Sandras" ;tag=b8aec11c-2104-da11-980a-00c09f2d8163 Contact: Call-ID: 68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 172.17.100.222:5080 itbx*CLI> -- Started music on hold, class 'default', on SIP/6005-5184 set_destination: Parsing for address/port to send to set_destination: set destination to 172.17.100.222, port 5080 We're at 172.17.100.111 port 10030 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x400 (ilbc) Answering with preferred capability 0x2 (gsm) Answering with non-codec capability 0x1 (telephone-event) 11 headers, 13 lines Reliably Transmitting: INVITE sip:6005@172.17.100.222:5080 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK2d712641;rport From: ;tag=as64a2cdf9 To: "Damien Sandras" ;tag=b8aec11c-2104-da11-980a-00c09f2d8163 Contact: Call-ID: 68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 293 v=0 o=root 24070 24072 IN IP4 172.17.100.111 s=session c=IN IP4 172.17.100.111 t=0 0 m=audio 10030 RTP/AVP 0 8 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 172.17.100.222:5080 itbx*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 172.17.100.222:5022 Found description format PCMU Found description format telephone-event Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 172.17.100.222, port 5080 Transmitting: ACK sip:6005@172.17.100.222:5080 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK4297b547;rport From: ;tag=as64a2cdf9 To: "Damien Sandras" ;tag=b8aec11c-2104-da11-980a-00c09f2d8163 Contact: Call-ID: 68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 172.17.100.222:5080 itbx*CLI> 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:6005@172.17.100.222:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK1dd5b774 From: "ITBX" ;tag=as51ae707c To: Contact: Call-ID: 00c4be4d59dc9cdb1b6dbf1d3dff0846@172.17.100.111 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Fri, 05 Aug 2005 13:18:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 172.17.100.222:5080 itbx*CLI> Sip read: SIP/2.0 200 OK CSeq: 102 OPTIONS Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK1dd5b774 From: "ITBX" ;tag=as51ae707c Call-ID: 00c4be4d59dc9cdb1b6dbf1d3dff0846@172.17.100.111 To: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, SUBSCRIBE, NOTIFY, REFER, MESSAGE Content-Length: 0 8 headers, 0 lines Destroying call '00c4be4d59dc9cdb1b6dbf1d3dff0846@172.17.100.111' itbx*CLI> -- Stopped music on hold on SIP/6005-5184 set_destination: Parsing for address/port to send to set_destination: set destination to 172.17.100.222, port 5080 We're at 172.17.100.111 port 10030 Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 11 headers, 10 lines Reliably Transmitting: INVITE sip:6005@172.17.100.222:5080 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK6a27b3ba;rport From: ;tag=as64a2cdf9 To: "Damien Sandras" ;tag=b8aec11c-2104-da11-980a-00c09f2d8163 Contact: Call-ID: 68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 220 v=0 o=root 24070 24073 IN IP4 172.17.100.248 s=session c=IN IP4 172.17.100.248 t=0 0 m=audio 50000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 172.17.100.222:5080 itbx*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 172.17.100.222:5022 Found description format PCMU Found description format telephone-event Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 172.17.100.222, port 5080 Transmitting: ACK sip:6005@172.17.100.222:5080 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK44df4611;rport From: ;tag=as64a2cdf9 To: "Damien Sandras" ;tag=b8aec11c-2104-da11-980a-00c09f2d8163 Contact: Call-ID: 68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01 CSeq: 104 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 172.17.100.222:5080 itbx*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 172.17.100.222, port 5080 We're at 172.17.100.111 port 10030 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x400 (ilbc) Answering with preferred capability 0x2 (gsm) Answering with non-codec capability 0x1 (telephone-event) 11 headers, 13 lines Reliably Transmitting: INVITE sip:6005@172.17.100.222:5080 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK7002e927;rport From: ;tag=as64a2cdf9 To: "Damien Sandras" ;tag=b8aec11c-2104-da11-980a-00c09f2d8163 Contact: Call-ID: 68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 293 v=0 o=root 24070 24074 IN IP4 172.17.100.111 s=session c=IN IP4 172.17.100.111 t=0 0 m=audio 10030 RTP/AVP 0 8 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 172.17.100.222:5080 -- Executing Macro("SIP/6005-5184", "hangup") in new stack -- Executing Hangup("SIP/6005-5184", "") in new stack itbx*CLI> Retransmitting #1 (no NAT): INVITE sip:6005@172.17.100.222:5080 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK7002e927;rport From: ;tag=as64a2cdf9 To: "Damien Sandras" ;tag=b8aec11c-2104-da11-980a-00c09f2d8163 Contact: Call-ID: 68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 293 v=0 o=root 24070 24074 IN IP4 172.17.100.111 s=session c=IN IP4 172.17.100.111 t=0 0 m=audio 10030 RTP/AVP 0 8 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 172.17.100.222:5080 itbx*CLI> Retransmitting #2 (no NAT): INVITE sip:6005@172.17.100.222:5080 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK7002e927;rport From: ;tag=as64a2cdf9 To: "Damien Sandras" ;tag=b8aec11c-2104-da11-980a-00c09f2d8163 Contact: Call-ID: 68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 293 v=0 o=root 24070 24074 IN IP4 172.17.100.111 s=session c=IN IP4 172.17.100.111 t=0 0 m=audio 10030 RTP/AVP 0 8 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 172.17.100.222:5080 itbx*CLI> Retransmitting #3 (no NAT): INVITE sip:6005@172.17.100.222:5080 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK7002e927;rport From: ;tag=as64a2cdf9 To: "Damien Sandras" ;tag=b8aec11c-2104-da11-980a-00c09f2d8163 Contact: Call-ID: 68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 293 v=0 o=root 24070 24074 IN IP4 172.17.100.111 s=session c=IN IP4 172.17.100.111 t=0 0 m=audio 10030 RTP/AVP 0 8 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 172.17.100.222:5080 itbx*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 172.17.100.222:5022 Found description format PCMU Found description format telephone-event Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 172.17.100.222, port 5080 Transmitting: ACK sip:6005@172.17.100.222:5080 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK2fdea358;rport From: ;tag=as64a2cdf9 To: "Damien Sandras" ;tag=b8aec11c-2104-da11-980a-00c09f2d8163 Contact: Call-ID: 68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01 CSeq: 105 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 172.17.100.222:5080 set_destination: Parsing for address/port to send to set_destination: set destination to 172.17.100.222, port 5080 Reliably Transmitting: BYE sip:6005@172.17.100.222:5080 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK45d32e44;rport From: ;tag=as64a2cdf9 To: "Damien Sandras" ;tag=b8aec11c-2104-da11-980a-00c09f2d8163 Contact: Call-ID: 68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01 CSeq: 106 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 172.17.100.222:5080 itbx*CLI> Destroying call '68a4c11c-2104-da11-980a-00c09f2d8163@golgoth01' itbx*CLI> itbx:/usr/src# Script done on ven 05 aoû 2005 15:18:16 CEST