Script started on ven 05 aoû 2005 15:12:43 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 = 23940) itbx*CLI> Verbosity is at least 4 Core debug is at least 4 itbx*CLI> -- Executing SetVar("SIP/6005-6ef4", "CALLEDID=6001") in new stack -- Executing GotoIf("SIP/6005-6ef4", "0?3:7") in new stack -- Goto (from-ip-phones,6001,7) -- Executing Macro("SIP/6005-6ef4", "dialuser|6001|30|rF") in new stack -- Executing DBget("SIP/6005-6ef4", "temp=FM/6001") in new stack -- DBget: varname=temp, family=FM, key=6001 -- DBget: set variable temp to 6001 -- Executing GotoIf("SIP/6005-6ef4", "1?102:3") in new stack -- Goto (macro-dialuser,s,102) itbx*CLI> -- Executing Dial("SIP/6005-6ef4", "SIP/6001|30|rF") in new stack We're at 172.17.100.111 port 12584 Answering/Requesting with root 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) 12 headers, 13 lines Reliably Transmitting: INVITE sip:6001@172.17.100.248:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK5d3508c1 From: "Damien 5" ;tag=as686a9f98 To: Contact: Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 102 INVITE User-Agent: ITBX Date: Fri, 05 Aug 2005 13:13:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 293 v=0 o=root 23941 23941 IN IP4 172.17.100.111 s=session c=IN IP4 172.17.100.111 t=0 0 m=audio 12584 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 172.17.100.248:5060 -- Called 6001 itbx*CLI> Sip read: SIP/2.0 180 Ringing From: "Damien 5";tag=as686a9f98 To: ;tag=f86411ac-13c4-2f27a-b830208-4a1 Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 102 INVITE Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK5d3508c1 Supported: replaces,100rel,timer Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS User-Agent: Swissvoice IP10S Contact: Content-Length: 0 11 headers, 0 lines -- SIP/6001-6510 is ringing itbx*CLI> Sip read: SIP/2.0 200 OK From: "Damien 5";tag=as686a9f98 To: ;tag=f86411ac-13c4-2f27a-b830208-4a1 Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 102 INVITE Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK5d3508c1 Supported: replaces,100rel,timer Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS User-Agent: Swissvoice IP10S Contact: Session-Expires: 3600;refresher=uas Content-Type: application/sdp Content-Length: 221 v=0 o=rtp/1 193145 193145 IN IP4 172.17.100.248 s=- c=IN IP4 172.17.100.248 t=0 0 m=audio 50000 RTP/AVP 0 101 a=ptime:30 a=SilenceSupp:off a=fmtp:101 0-15 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 13 headers, 11 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 172.17.100.248:50000 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) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 172.17.100.248, port 5060 Transmitting: ACK sip:6001@172.17.100.248:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK6be51dc4 From: "Damien 5" ;tag=as686a9f98 To: ;tag=f86411ac-13c4-2f27a-b830208-4a1 Contact: Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 102 ACK User-Agent: ITBX Content-Length: 0 (no NAT) to 172.17.100.248:5060 -- SIP/6001-6510 answered SIP/6005-6ef4 -- Attempting native bridge of SIP/6005-6ef4 and SIP/6001-6510 set_destination: Parsing for address/port to send to set_destination: set destination to 172.17.100.248, port 5060 We're at 172.17.100.111 port 12584 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 11 headers, 11 lines Reliably Transmitting: INVITE sip:6001@172.17.100.248:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK30eacc4d From: "Damien 5" ;tag=as686a9f98 To: ;tag=f86411ac-13c4-2f27a-b830208-4a1 Contact: Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 103 INVITE User-Agent: ITBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 243 v=0 o=root 23941 23942 IN IP4 172.17.100.222 s=session c=IN IP4 172.17.100.222 t=0 0 m=audio 5020 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 172.17.100.248:5060 itbx*CLI> Sip read: SIP/2.0 200 OK From: "Damien 5";tag=as686a9f98 To: ;tag=f86411ac-13c4-2f27a-b830208-4a1 Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 103 INVITE Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK30eacc4d Supported: replaces,100rel,timer User-Agent: Swissvoice IP10S Contact: Content-Type: application/sdp Content-Length: 203 v=0 o=6001 193148 193148 IN IP4 172.17.100.248 s=- c=IN IP4 172.17.100.248 t=0 0 m=audio 50000 RTP/AVP 0 101 a=ptime:30 a=SilenceSupp:off a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 11 headers, 10 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 172.17.100.248:50000 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 172.17.100.248, port 5060 Transmitting: ACK sip:6001@172.17.100.248:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK0d41433c From: "Damien 5" ;tag=as686a9f98 To: ;tag=f86411ac-13c4-2f27a-b830208-4a1 Contact: Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 103 ACK User-Agent: ITBX Content-Length: 0 (no NAT) to 172.17.100.248:5060 itbx*CLI> Sip read: INVITE sip:6005@172.17.100.111 SIP/2.0 From: ;tag=f86411ac-13c4-2f27a-b830208-4a1 To: "Damien 5";tag=as686a9f98 Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 1 INVITE Via: SIP/2.0/UDP 172.17.100.248:5060;rport;branch=z9hG4bK-2f27f-b83153e-29b4 Max-Forwards: 70 Supported: replaces,100rel,timer User-Agent: Swissvoice IP10S Contact: Content-Type: application/sdp Content-Length: 208 itbx*CLI> v=0 o=6001 193148 193149 IN IP4 172.17.100.248 s=- c=IN IP4 0.0.0.0 t=0 0 m=audio 50000 RTP/AVP 0 101 a=ptime:30 a=SilenceSupp:off a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=sendonly 12 headers, 11 lines Using latest request as basis request Sending to 172.17.100.248 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:50000 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) -- Started music on hold, class 'default', on SIP/6005-6ef4 We're at 172.17.100.111 port 12584 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.100.248:5060;branch=z9hG4bK-2f27f-b83153e-29b4 From: ;tag=f86411ac-13c4-2f27a-b830208-4a1 To: "Damien 5";tag=as686a9f98 Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 1 INVITE User-Agent: ITBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 23941 23943 IN IP4 172.17.100.222 s=session c=IN IP4 172.17.100.222 t=0 0 m=audio 5020 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 172.17.100.248:5060 itbx*CLI> Sip read: ACK sip:6005@172.17.100.111 SIP/2.0 From: ;tag=f86411ac-13c4-2f27a-b830208-4a1 To: "Damien 5";tag=as686a9f98 Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 1 ACK Via: SIP/2.0/UDP 172.17.100.248:5060;rport;branch=z9hG4bK-2f27f-b831664-49dd Max-Forwards: 70 User-Agent: Swissvoice IP10S Contact: Content-Length: 0 10 headers, 0 lines itbx*CLI> Sip read: INVITE sip:6005@172.17.100.111 SIP/2.0 From: ;tag=f86411ac-13c4-2f27a-b830208-4a1 To: "Damien 5";tag=as686a9f98 Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 2 INVITE Via: SIP/2.0/UDP 172.17.100.248:5060;rport;branch=z9hG4bK-2f285-b832dce-5423 Max-Forwards: 70 Supported: replaces,100rel,timer User-Agent: Swissvoice IP10S Contact: Content-Type: application/sdp Content-Length: 215 itbx*CLI> v=0 o=6001 193148 193150 IN IP4 172.17.100.248 s=- c=IN IP4 172.17.100.248 t=0 0 m=audio 50000 RTP/AVP 0 101 a=ptime:30 a=SilenceSupp:off a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 12 headers, 11 lines Using latest request as basis request Sending to 172.17.100.248 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 172.17.100.248:50000 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) -- Stopped music on hold on SIP/6005-6ef4 We're at 172.17.100.111 port 12584 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.100.248:5060;branch=z9hG4bK-2f285-b832dce-5423 From: ;tag=f86411ac-13c4-2f27a-b830208-4a1 To: "Damien 5";tag=as686a9f98 Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 2 INVITE User-Agent: ITBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 23941 23944 IN IP4 172.17.100.222 s=session c=IN IP4 172.17.100.222 t=0 0 m=audio 5020 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 172.17.100.248:5060 itbx*CLI> Sip read: ACK sip:6005@172.17.100.111 SIP/2.0 From: ;tag=f86411ac-13c4-2f27a-b830208-4a1 To: "Damien 5";tag=as686a9f98 Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 2 ACK Via: SIP/2.0/UDP 172.17.100.248:5060;rport;branch=z9hG4bK-2f285-b832f10-2d20 Max-Forwards: 70 User-Agent: Swissvoice IP10S Contact: Content-Length: 0 10 headers, 0 lines itbx*CLI> Sip read: BYE sip:6005@172.17.100.111 SIP/2.0 From: ;tag=f86411ac-13c4-2f27a-b830208-4a1 To: "Damien 5";tag=as686a9f98 Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 3 BYE Via: SIP/2.0/UDP 172.17.100.248:5060;rport;branch=z9hG4bK-2f28a-b834366-589e Max-Forwards: 70 Supported: replaces,100rel,timer User-Agent: Swissvoice IP10S Content-Length: 0 10 headers, 0 lines Sending to 172.17.100.248 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.100.248:5060;branch=z9hG4bK-2f28a-b834366-589e From: ;tag=f86411ac-13c4-2f27a-b830208-4a1 To: "Damien 5";tag=as686a9f98 Call-ID: 1cd7720548b164750158c23638312abd@172.17.100.111 CSeq: 3 BYE User-Agent: ITBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 172.17.100.248:5060 -- Executing Macro("SIP/6005-6ef4", "hangup") in new stack -- Executing Hangup("SIP/6005-6ef4", "") in new stack itbx*CLI> Destroying call '1cd7720548b164750158c23638312abd@172.17.100.111' itbx*CLI> 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:6001@172.17.100.248:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK05c78ac8 From: "ITBX" ;tag=as5b9cf372 To: Contact: Call-ID: 533430492ab52a0b238aaa1710dfbfbb@172.17.100.111 CSeq: 102 OPTIONS User-Agent: ITBX Date: Fri, 05 Aug 2005 13:13:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 172.17.100.248:5060 itbx*CLI> Sip read: SIP/2.0 200 OK From: "ITBX";tag=as5b9cf372 To: ;tag=f86411ac-13c4-2f28b-b8346a0-7b4c Call-ID: 533430492ab52a0b238aaa1710dfbfbb@172.17.100.111 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 172.17.100.111:5060;branch=z9hG4bK05c78ac8 Supported: replaces,100rel,timer Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS User-Agent: Swissvoice IP10S Accept: application/sdp Content-Length: 0 11 headers, 0 lines Destroying call '533430492ab52a0b238aaa1710dfbfbb@172.17.100.111' itbx*CLI> itbx:/usr/src# Script done on ven 05 aoû 2005 15:13:27 CEST