Asterisk 1.2.12.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.12.1 currently running on ez-alb-voip1 (pid = 26608) Verbosity is at least 10 Core debug is at least 10 ez-alb-voip1*CLI> sip debug SIP Debugging enabled ez-alb-voip1*CLI> <-- SIP read from 4.79.xxx.xxx:5060: INVITE sip:+1npanxxxxxx@65.90.xx.xx:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 4.79.xxx.xxx;branch=z9hG4bKac5e.34f92501.0 Via: SIP/2.0/UDP 4.68.xxx.xxx:5060;branch=z9hG4bK506071629460-1158244926856 From: "Name Witheld" ;tag=VPSF506071629460 To: Call-ID: BOSMGC0120061003141558019859@209.244.xx.xx CSeq: 1 INVITE Contact: Max-Forwards: 68 Content-Type: application/sdp Content-Length: 173 Remote-Party-ID: "Name Witheld" ;party=calling;screen=no;privacy=off v=0 o=- 1159884958 1159884959 IN IP4 209.247.xxx.xxx s=- c=IN IP4 209.247.xxx.xxx t=0 0 m=audio 60146 RTP/AVP 0 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (13 headers 8 lines)--- Using INVITE request as basis request - BOSMGC0120061003141558019859@209.244.xxx.xxx Sending to 4.79.xxx.xxx : 5060 (non-NAT) Found peer 'bw-out-testing' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 209.247.xxx.xxx:60146 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for +1npanxxxxxx in from-trunk (domain 65.90.xxx.xxx) list_route: hop: Transmitting (no NAT) to 4.79.xxx.xxx:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 4.79.xxx.xxx;branch=z9hG4bKac5e.34f92501.0;received=4.79.xxx.xxx Via: SIP/2.0/UDP 4.68.xxx.xxx:5060;branch=z9hG4bK506071629460-1158244926856 From: "Name Witheld" ;tag=VPSF506071629460 To: Call-ID: BOSMGC0120061003141558019859@209.244.xxx.xxx CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Set("SIP/4.68.xxx.xxx-b7005f68", "FROM_DID=+1npanxxxxxx") in new stack -- Executing Goto("SIP/4.68.xxx.xxx-b7005f68", "ext-queues|1060|1") in new stack -- Goto (ext-queues,1060,1) -- Executing Answer("SIP/4.68.xxx.xxx-b7005f68", "") in new stack We're at 65.90.xxx.xxx port 19334 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 4.79.xxx.xxx:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 4.79.xxx.xxx;branch=z9hG4bKac5e.34f92501.0;received=4.79.xxx.xxx Via: SIP/2.0/UDP 4.68.xxx.xxx:5060;branch=z9hG4bK506071629460-1158244926856 Record-Route: From: "Name Witheld" ;tag=VPSF506071629460 To: ;tag=as69378d2f Call-ID: BOSMGC0120061003141558019859@209.244.xxx.xxx CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 26608 26608 IN IP4 65.90.xxx.xxx s=session c=IN IP4 65.90.xxx.xxx t=0 0 m=audio 19334 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing GotoIf("SIP/4.68.xxx.xxx-b7005f68", "0?USERCID:SETCID") in new stack -- Goto (ext-queues,1060,4) -- Executing Set("SIP/4.68.xxx.xxx-b7005f68", "CALLERID(name)=alb:Name Witheld") in new stack -- Executing Set("SIP/4.68.xxx.xxx-b7005f68", "MONITOR_FILENAME=/var/spool/asterisk/monitor/q1060-20061003-101558-1159884958.95") in new stack -- Executing Playback("SIP/4.68.xxx.xxx-b7005f68", "custom/TYFC-Generic") in new stack -- Playing 'custom/TYFC-Generic' (language 'en') ez-alb-voip1*CLI> <-- SIP read from 4.79.xxx.xxx:5060: ACK sip:+1npanxxxxxx@65.90.xxx.xxx SIP/2.0 Record-Route: Via: SIP/2.0/UDP 4.79.xxx.xxx;branch=0 Via: SIP/2.0/UDP 4.68.xxx.xxx:5060;branch=z9hG4bK506071629460-1158244926864 From: "Name Witheld" ;tag=VPSF506071629460 To: ;tag=as69378d2f Call-ID: BOSMGC0120061003141558019859@209.244.xxx.xxx CSeq: 1 ACK Contact: Max-Forwards: 69 Content-Length: 0 --- (11 headers 0 lines)--- -- Executing Queue("SIP/4.68.xxx.xxx-b7005f68", "1060|t|||180") in new stack -- Started music on hold, class 'MOHDisplayCenter', on SIP/4.68.xxx.xxx-b7005f68 -- Called Local/1011@from-internal/n -- Called Local/1013@from-internal/n -- Executing Macro("Local/1011@from-internal-7886,2", "exten-vm|1011|1011") in new stack -- Executing Macro("Local/1013@from-internal-bdd5,2", "exten-vm|1013|1013") in new stack -- Called Local/1012@from-internal/n -- Executing Macro("Local/1013@from-internal-bdd5,2", "user-callerid") in new stack -- Executing Macro("Local/1011@from-internal-7886,2", "user-callerid") in new stack -- Executing Macro("Local/1012@from-internal-8ab0,2", "exten-vm|1012|1012") in new stack -- Executing GotoIf("Local/1011@from-internal-7886,2", "1?report") in new stack -- Goto (macro-user-callerid,s,9) -- Executing Macro("Local/1012@from-internal-8ab0,2", "user-callerid") in new stack -- Executing GotoIf("Local/1013@from-internal-bdd5,2", "1?report") in new stack -- Goto (macro-user-callerid,s,9) -- Executing NoOp("Local/1011@from-internal-7886,2", "Using CallerID "alb:Name Witheld" <+1npanxxxxxx>") in new stack -- Executing GotoIf("Local/1012@from-internal-8ab0,2", "1?report") in new stack -- Goto (macro-user-callerid,s,9) -- Executing NoOp("Local/1012@from-internal-8ab0,2", "Using CallerID "alb:Name Witheld" <+1npanxxxxxx>") in new stack -- Executing Set("Local/1012@from-internal-8ab0,2", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("Local/1012@from-internal-8ab0,2", "VMBOX=1012") in new stack -- Executing Set("Local/1012@from-internal-8ab0,2", "EXTTOCALL=1012") in new stack -- Executing Set("Local/1011@from-internal-7886,2", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("Local/1011@from-internal-7886,2", "VMBOX=1011") in new stack -- Executing Set("Local/1011@from-internal-7886,2", "EXTTOCALL=1011") in new stack -- Executing Set("Local/1011@from-internal-7886,2", "CFUEXT=") in new stack -- Executing Set("Local/1012@from-internal-8ab0,2", "CFUEXT=") in new stack -- Executing NoOp("Local/1013@from-internal-bdd5,2", "Using CallerID "alb:Name Witheld" <+1npanxxxxxx>") in new stack -- Executing Set("Local/1013@from-internal-bdd5,2", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("Local/1011@from-internal-7886,2", "RT=20") in new stack -- Executing Set("Local/1013@from-internal-bdd5,2", "VMBOX=1013") in new stack -- Executing Macro("Local/1011@from-internal-7886,2", "record-enable|1011|IN") in new stack -- Executing Set("Local/1012@from-internal-8ab0,2", "RT=20") in new stack -- Executing Set("Local/1013@from-internal-bdd5,2", "EXTTOCALL=1013") in new stack -- Executing GotoIf("Local/1011@from-internal-7886,2", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("Local/1011@from-internal-7886,2", "recordingcheck|20061003-101604|1159884964.97") in new stack -- Executing Macro("Local/1012@from-internal-8ab0,2", "record-enable|1012|IN") in new stack -- Executing Set("Local/1013@from-internal-bdd5,2", "CFUEXT=") in new stack -- Executing GotoIf("Local/1012@from-internal-8ab0,2", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- Executing Set("Local/1013@from-internal-bdd5,2", "RT=20") in new stack -- Executing Macro("Local/1013@from-internal-bdd5,2", "record-enable|1013|IN") in new stack -- Executing GotoIf("Local/1013@from-internal-bdd5,2", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("Local/1013@from-internal-bdd5,2", "recordingcheck|20061003-101604|1159884964.99") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- Executing AGI("Local/1012@from-internal-8ab0,2", "recordingcheck|20061003-101604|1159884964.101") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20061003-101604|1159884964.101: Inbound recording enabled. recordingcheck|20061003-101604|1159884964.101: CALLFILENAME=20061003-101604-1159884964.101 -- AGI Script recordingcheck completed, returning 0 -- Executing Monitor("Local/1012@from-internal-8ab0,2", "wav49|20061003-101604-1159884964.101| mb") in new stack -- Executing GotoIf("Local/1012@from-internal-8ab0,2", "1?dolocaldial|1") in new stack -- Goto (macro-exten-vm,dolocaldial,1) -- Executing Macro("Local/1012@from-internal-8ab0,2", "dial||tr|1012") in new stack -- Executing AGI("Local/1012@from-internal-8ab0,2", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi recordingcheck|20061003-101604|1159884964.99: Inbound recording enabled. recordingcheck|20061003-101604|1159884964.99: CALLFILENAME=20061003-101604-1159884964.99 -- AGI Script recordingcheck completed, returning 0 -- Executing Monitor("Local/1013@from-internal-bdd5,2", "wav49|20061003-101604-1159884964.99| mb") in new stack -- Executing GotoIf("Local/1013@from-internal-bdd5,2", "1?dolocaldial|1") in new stack -- Goto (macro-exten-vm,dolocaldial,1) -- Executing Macro("Local/1013@from-internal-bdd5,2", "dial||tr|1013") in new stack -- Executing AGI("Local/1013@from-internal-bdd5,2", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi recordingcheck|20061003-101604|1159884964.97: Inbound recording enabled. recordingcheck|20061003-101604|1159884964.97: CALLFILENAME=20061003-101604-1159884964.97 -- AGI Script recordingcheck completed, returning 0 -- Executing Monitor("Local/1011@from-internal-7886,2", "wav49|20061003-101604-1159884964.97| mb") in new stack -- Executing GotoIf("Local/1011@from-internal-7886,2", "1?dolocaldial|1") in new stack -- Goto (macro-exten-vm,dolocaldial,1) -- Executing Macro("Local/1011@from-internal-7886,2", "dial||tr|1011") in new stack -- Executing AGI("Local/1011@from-internal-7886,2", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi -- dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'alb:Name Witheld' number is '+1npanxxxxxx' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 1012 to extension map -- dialparties.agi: Extension 1012 cf is disabled -- dialparties.agi: Extension 1012 do not disturb is disabled > dialparties.agi: extnum: 1012 > dialparties.agi: exthascw: 0 > dialparties.agi: exthascfb: 0 > dialparties.agi: extcfb: > dialparties.agi: exthascfu: 0 > dialparties.agi: extcfu: == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 > dialparties.agi: ExtensionState: -1 -- dialparties.agi: Checking CW and CFB status for extension 1012 dialparties.agi: Starting New Dialparties.agi -- dialparties.agi: priority is 1 -- dialparties.agi: DbSet CALLTRACE/1012 to +1npanxxxxxx -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("Local/1012@from-internal-8ab0,2", "SIP/1030||tr") in new stack We're at 192.168.1.50 port 10630 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 192.168.1.201:5060: INVITE sip:1030@192.168.1.201:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK6370e529 From: "alb:Name Witheld" ;tag=as47d1ab40 To: Contact: Call-ID: 33d4fc8a73828a03105159e94d272da5@192.168.1.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 03 Oct 2006 14:16:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 26608 26608 IN IP4 192.168.1.50 s=session c=IN IP4 192.168.1.50 t=0 0 m=audio 10630 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 - - - - --- -- Called 1030 -- Local/1012@from-internal-8ab0,1 is ringing dialparties.agi: Caller ID name is 'alb:Name Witheld' number is '+1npanxxxxxx' dialparties.agi: Starting New Dialparties.agi dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: priority is 1 -- dialparties.agi: Added extension 1013 to extension map dialparties.agi: Caller ID name is 'alb:Name Witheld' number is '+1npanxxxxxx' -- dialparties.agi: Extension 1013 cf is disabled dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Extension 1013 do not disturb is disabled -- dialparties.agi: Added extension 1011 to extension map > dialparties.agi: extnum: 1013 -- dialparties.agi: Extension 1011 cf is disabled > dialparties.agi: exthascw: 0 -- dialparties.agi: Extension 1011 do not disturb is disabled > dialparties.agi: exthascfb: 0 > dialparties.agi: extnum: 1011 > dialparties.agi: extcfb: > dialparties.agi: exthascw: 0 > dialparties.agi: exthascfu: 0 > dialparties.agi: exthascfb: 0 > dialparties.agi: extcfu: > dialparties.agi: extcfb: > dialparties.agi: exthascfu: 0 > dialparties.agi: ExtensionState: -1 > dialparties.agi: extcfu: -- dialparties.agi: Checking CW and CFB status for extension 1013 > dialparties.agi: ExtensionState: -1 -- dialparties.agi: DbSet CALLTRACE/1013 to +1npanxxxxxx -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("Local/1013@from-internal-bdd5,2", "SIP/1032||tr") in new stack We're at 192.168.1.50 port 18242 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 192.168.1.204:5060: INVITE sip:1032@192.168.1.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK6e44c8b8 From: "alb:Name Witheld" ;tag=as08b68b29 To: Contact: Call-ID: 349c5f1a734a9668241e473c3fffc67d@192.168.1.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 03 Oct 2006 14:16:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 26608 26608 IN IP4 192.168.1.50 s=session c=IN IP4 192.168.1.50 t=0 0 m=audio 18242 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 - - - - --- -- Called 1032 -- Local/1013@from-internal-bdd5,1 is ringing -- dialparties.agi: Checking CW and CFB status for extension 1011 -- dialparties.agi: DbSet CALLTRACE/1011 to +1npanxxxxxx -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("Local/1011@from-internal-7886,2", "SIP/1031||tr") in new stack We're at 192.168.1.50 port 10862 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 192.168.1.205:5060: INVITE sip:1031@192.168.1.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK3ad405fe From: "alb:Name Witheld" ;tag=as7fdbf8b7 To: Contact: Call-ID: 24f1ca51131e9b3e28bb1427594d8742@192.168.1.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 03 Oct 2006 14:16:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 26608 26608 IN IP4 192.168.1.50 s=session c=IN IP4 192.168.1.50 t=0 0 m=audio 10862 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 - - - - --- -- Called 1031 -- Local/1011@from-internal-7886,1 is ringing ez-alb-voip1*CLI> <-- SIP read from 192.168.1.201:50052: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK6370e529 From: "alb:Name Witheld" ;tag=as47d1ab40 To: Call-ID: 33d4fc8a73828a03105159e94d272da5@192.168.1.50 Date: Tue, 03 Oct 2006 GMT CSeq: 102 INVITE Server: Cisco-CP7961G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "1030" ;party=called;id-type=subscriber;privacy=off;screen=yes Allow-Events: kpml,dialog Content-Length: 0 --- (13 headers 0 lines)--- ez-alb-voip1*CLI> <-- SIP read from 192.168.1.205:50232: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK3ad405fe From: "alb:Name Witheld" ;tag=as7fdbf8b7 To: Call-ID: 24f1ca51131e9b3e28bb1427594d8742@192.168.1.50 Date: Tue, 03 Oct 2006 GMT CSeq: 102 INVITE Server: Cisco-CP7961G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "1031" ;party=called;id-type=subscriber;privacy=off;screen=yes Allow-Events: kpml,dialog Content-Length: 0 --- (13 headers 0 lines)--- ez-alb-voip1*CLI> <-- SIP read from 192.168.1.204:52168: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK6e44c8b8 From: "alb:Name Witheld" ;tag=as08b68b29 To: Call-ID: 349c5f1a734a9668241e473c3fffc67d@192.168.1.50 Date: Tue, 03 Oct 2006 GMT CSeq: 102 INVITE Server: Cisco-CP7961G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "1032" ;party=called;id-type=subscriber;privacy=off;screen=yes Allow-Events: kpml,dialog Content-Length: 0 --- (13 headers 0 lines)--- ez-alb-voip1*CLI> <-- SIP read from 192.168.1.205:50235: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK3ad405fe From: "alb:Name Witheld" ;tag=as7fdbf8b7 To: ;tag=00179405c65e005b17e72d7f-597e4120 Call-ID: 24f1ca51131e9b3e28bb1427594d8742@192.168.1.50 Date: Tue, 03 Oct 2006 GMT CSeq: 102 INVITE Server: Cisco-CP7961G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "1031" ;party=called;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Allow-Events: kpml,dialog Content-Length: 204 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 94 0 IN IP4 192.168.1.205 s=SIP Call t=0 0 m=audio 30662 RTP/AVP 0 101 c=IN IP4 192.168.1.205 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (16 headers 10 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.205:30662 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.205, port 5060 Transmitting (no NAT) to 192.168.1.205:5060: ACK sip:1031@192.168.1.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK07703d97 From: "alb:Name Witheld" ;tag=as7fdbf8b7 To: ;tag=00179405c65e005b17e72d7f-597e4120 Contact: Call-ID: 24f1ca51131e9b3e28bb1427594d8742@192.168.1.50 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/1031-b6c10668 answered Local/1011@from-internal-7886,2 -- Local/1011@from-internal-7886,1 answered SIP/4.68.xxx.xxx-b7005f68 -- Stopped music on hold on SIP/4.68.xxx.xxx-b7005f68 Scheduling destruction of call '349c5f1a734a9668241e473c3fffc67d@192.168.1.50' in 32000 ms Scheduling destruction of call '33d4fc8a73828a03105159e94d272da5@192.168.1.50' in 32000 ms ez-alb-voip1*CLI> <-- SIP read from 4.79.xxx.xxx:5060: BYE sip:+1npanxxxxxx@65.90.xxx.xxx SIP/2.0 Record-Route: Via: SIP/2.0/UDP 4.79.xxx.xxx;branch=z9hG4bK7c5e.b6f47ef4.0 Via: SIP/2.0/UDP 4.68.xxx.xxx:5060;branch=z9hG4bK506071629460-1158244928124 From: "Name Witheld" ;tag=VPSF506071629460 To: ;tag=as69378d2f Call-ID: BOSMGC0120061003141558019859@209.244.xxx.xxx CSeq: 2 BYE Contact: Max-Forwards: 68 Content-Length: 0 --- (11 headers 0 lines)--- Sending to 4.79.xxx.xxx : 5060 (non-NAT) Transmitting (no NAT) to 4.79.xxx.xxx:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 4.79.xxx.xxx;branch=z9hG4bK7c5e.b6f47ef4.0;received=4.79.xxx.xxx Via: SIP/2.0/UDP 4.68.xxx.xxx:5060;branch=z9hG4bK506071629460-1158244928124 Record-Route: From: "Name Witheld" ;tag=VPSF506071629460 To: ;tag=as69378d2f Call-ID: BOSMGC0120061003141558019859@209.244.xxx.xxx CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Scheduling destruction of call '24f1ca51131e9b3e28bb1427594d8742@192.168.1.50' in 32000 ms set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.205, port 5060 Reliably Transmitting (no NAT) to 192.168.1.205:5060: BYE sip:1031@192.168.1.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK7bf85466 From: "alb:Name Witheld" ;tag=as7fdbf8b7 To: ;tag=00179405c65e005b17e72d7f-597e4120 Contact: Call-ID: 24f1ca51131e9b3e28bb1427594d8742@192.168.1.50 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Retransmitting #1 (no NAT) to 192.168.1.205:5060: BYE sip:1031@192.168.1.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK7bf85466 From: "alb:Name Witheld" ;tag=as7fdbf8b7 To: ;tag=00179405c65e005b17e72d7f-597e4120 Contact: Call-ID: 24f1ca51131e9b3e28bb1427594d8742@192.168.1.50 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Destroying call 'BOSMGC0120061003141558019859@209.244.xxx.xxx' Retransmitting #2 (no NAT) to 192.168.1.205:5060: BYE sip:1031@192.168.1.205:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK7bf85466 From: "alb:Name Witheld" ;tag=as7fdbf8b7 To: ;tag=00179405c65e005b17e72d7f-597e4120 Contact: Call-ID: 24f1ca51131e9b3e28bb1427594d8742@192.168.1.50 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- ez-alb-voip1*CLI> <-- SIP read from 192.168.1.205:50239: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK7bf85466 From: "alb:Name Witheld" ;tag=as7fdbf8b7 To: ;tag=00179405c65e005b17e72d7f-597e4120 Call-ID: 24f1ca51131e9b3e28bb1427594d8742@192.168.1.50 Date: Tue, 03 Oct 2006 GMT CSeq: 103 BYE Server: Cisco-CP7961G/8.0 Content-Length: 0 RTP-RxStat: Dur=12,Pkt=597,Oct=102684,LatePkt=0,LostPkt=0,AvgJit=0 RTP-TxStat: Dur=12,Pkt=597,Oct=102684 --- (11 headers 0 lines)--- Destroying call '24f1ca51131e9b3e28bb1427594d8742@192.168.1.50' ez-alb-voip1*CLI> show channels Channel Location State Application(Data) 0 active channels 0 active calls ez-alb-voip1*CLI> <-- SIP read from 192.168.1.204:52171: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK6e44c8b8 From: "alb:Name Witheld" ;tag=as08b68b29 To: ;tag=00179405dc28006b5ed97d32-03189fc1 Call-ID: 349c5f1a734a9668241e473c3fffc67d@192.168.1.50 Date: Tue, 03 Oct 2006 GMT CSeq: 102 INVITE Server: Cisco-CP7961G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "1032" ;party=called;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Allow-Events: kpml,dialog Content-Length: 207 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 23605 0 IN IP4 192.168.1.204 s=SIP Call t=0 0 m=audio 17440 RTP/AVP 0 101 c=IN IP4 192.168.1.204 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (16 headers 10 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.204:17440 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.204, port 5060 Transmitting (no NAT) to 192.168.1.204:5060: ACK sip:1032@192.168.1.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK3c321420 From: "alb:Name Witheld" ;tag=as08b68b29 To: ;tag=00179405dc28006b5ed97d32-03189fc1 Contact: Call-ID: 349c5f1a734a9668241e473c3fffc67d@192.168.1.50 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.204, port 5060 Reliably Transmitting (no NAT) to 192.168.1.204:5060: BYE sip:1032@192.168.1.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK47266a91 From: "alb:Name Witheld" ;tag=as08b68b29 To: ;tag=00179405dc28006b5ed97d32-03189fc1 Contact: Call-ID: 349c5f1a734a9668241e473c3fffc67d@192.168.1.50 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '349c5f1a734a9668241e473c3fffc67d@192.168.1.50' in 32000 ms Retransmitting #1 (no NAT) to 192.168.1.204:5060: BYE sip:1032@192.168.1.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK47266a91 From: "alb:Name Witheld" ;tag=as08b68b29 To: ;tag=00179405dc28006b5ed97d32-03189fc1 Contact: Call-ID: 349c5f1a734a9668241e473c3fffc67d@192.168.1.50 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Retransmitting #2 (no NAT) to 192.168.1.204:5060: BYE sip:1032@192.168.1.204:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK47266a91 From: "alb:Name Witheld" ;tag=as08b68b29 To: ;tag=00179405dc28006b5ed97d32-03189fc1 Contact: Call-ID: 349c5f1a734a9668241e473c3fffc67d@192.168.1.50 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- ez-alb-voip1*CLI> <-- SIP read from 192.168.1.204:52175: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK47266a91 From: "alb:Name Witheld" ;tag=as08b68b29 To: ;tag=00179405dc28006b5ed97d32-03189fc1 Call-ID: 349c5f1a734a9668241e473c3fffc67d@192.168.1.50 Date: Tue, 03 Oct 2006 GMT CSeq: 103 BYE Server: Cisco-CP7961G/8.0 Content-Length: 0 RTP-RxStat: Dur=1159884990,Pkt=0,Oct=0,LatePkt=0,LostPkt=0,AvgJit=0 RTP-TxStat: Dur=0,Pkt=2,Oct=344 --- (11 headers 0 lines)--- Destroying call '349c5f1a734a9668241e473c3fffc67d@192.168.1.50' Destroying call '33d4fc8a73828a03105159e94d272da5@192.168.1.50' ez-alb-voip1*CLI> <-- SIP read from 192.168.1.201:50055: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK6370e529 From: "alb:Name Witheld" ;tag=as47d1ab40 To: ;tag=0017e0575d84006fd29dd037-ac567535 Call-ID: 33d4fc8a73828a03105159e94d272da5@192.168.1.50 Date: Tue, 03 Oct 2006 GMT CSeq: 102 INVITE Server: Cisco-CP7961G/8.0 Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "1030" ;party=called;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Allow-Events: kpml,dialog Content-Length: 206 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 4800 0 IN IP4 192.168.1.201 s=SIP Call t=0 0 m=audio 25974 RTP/AVP 0 101 c=IN IP4 192.168.1.201 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (16 headers 10 lines)--- Destroying call '33d4fc8a73828a03105159e94d272da5@192.168.1.50'