Connected to Asterisk 1.4.17-1 RPM by vc-rpms@voipconsulting.nl currently running on trixbox1 (pid = 2570) Verbosity is at least 12 == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 -- Executing [090222902@from-internal:1] Macro("SIP/1000-091f5490", "dialout-trunk|1|90222902||") in new stack -- Executing [s@macro-dialout-trunk:1] Set("SIP/1000-091f5490", "DIAL_TRUNK=1") in new stack -- Executing [s@macro-dialout-trunk:2] Set("SIP/1000-091f5490", "DIAL_NUMBER=90222902") in new stack -- Executing [s@macro-dialout-trunk:3] Set("SIP/1000-091f5490", "ROUTE_PASSWD=") in new stack -- Executing [s@macro-dialout-trunk:4] GotoIf("SIP/1000-091f5490", "1?noauth") in new stack -- Goto (macro-dialout-trunk,s,6) -- Executing [s@macro-dialout-trunk:6] GotoIf("SIP/1000-091f5490", "0?disabletrunk|1") in new stack -- Executing [s@macro-dialout-trunk:7] Set("SIP/1000-091f5490", "_NODEST=") in new stack -- Executing [s@macro-dialout-trunk:8] Set("SIP/1000-091f5490", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s@macro-dialout-trunk:9] Set("SIP/1000-091f5490", "GROUP()=OUT_1") in new stack -- Executing [s@macro-dialout-trunk:10] Macro("SIP/1000-091f5490", "user-callerid|SKIPTTL") in new stack -- Executing [s@macro-user-callerid:1] NoOp("SIP/1000-091f5490", "user-callerid: device 1000") in new stack -- Executing [s@macro-user-callerid:2] Set("SIP/1000-091f5490", "AMPUSER=1000") in new stack -- Executing [s@macro-user-callerid:3] GotoIf("SIP/1000-091f5490", "0?report") in new stack -- Executing [s@macro-user-callerid:4] GotoIf("SIP/1000-091f5490", "0?start") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/1000-091f5490", "REALCALLERIDNUM=1000") in new stack -- Executing [s@macro-user-callerid:6] NoOp("SIP/1000-091f5490", "REALCALLERIDNUM is 1000") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/1000-091f5490", "AMPUSER=1000") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/1000-091f5490", "AMPUSERCIDNAME=Nicolai Solem HOME") in new stack -- Executing [s@macro-user-callerid:9] GotoIf("SIP/1000-091f5490", "0?report") in new stack -- Executing [s@macro-user-callerid:10] Set("SIP/1000-091f5490", "AMPUSERCID=1000") in new stack -- Executing [s@macro-user-callerid:11] Set("SIP/1000-091f5490", "CALLERID(all)="Nicolai Solem HOME" <1000>") in new stack -- Executing [s@macro-user-callerid:12] Set("SIP/1000-091f5490", "REALCALLERIDNUM=1000") in new stack -- Executing [s@macro-user-callerid:13] NoOp("SIP/1000-091f5490", "TTL: ARG1: SKIPTTL") in new stack -- Executing [s@macro-user-callerid:14] GotoIf("SIP/1000-091f5490", "1?continue") in new stack -- Goto (macro-user-callerid,s,23) -- Executing [s@macro-user-callerid:23] NoOp("SIP/1000-091f5490", "Using CallerID "Nicolai Solem HOME" <1000>") in new stack -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1000-091f5490", "record-enable|1000|OUT") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/1000-091f5490", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] AGI("SIP/1000-091f5490", "recordingcheck|20080205-200906|1202238546.456") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20080205-200906|1202238546.456: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing [s@macro-record-enable:5] NoOp("SIP/1000-091f5490", "No recording needed") in new stack -- Executing [s@macro-dialout-trunk:12] GotoIf("SIP/1000-091f5490", "0?skipoutcid") in new stack -- Executing [s@macro-dialout-trunk:13] Set("SIP/1000-091f5490", "DIAL_TRUNK_OPTIONS=TW") in new stack -- Executing [s@macro-dialout-trunk:14] Macro("SIP/1000-091f5490", "outbound-callerid|1") in new stack -- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/1000-091f5490", "1?start") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing [s@macro-outbound-callerid:3] NoOp("SIP/1000-091f5490", "REALCALLERIDNUM is 1000") in new stack -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/1000-091f5490", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,9) -- Executing [s@macro-outbound-callerid:9] Set("SIP/1000-091f5490", "USEROUTCID=") in new stack -- Executing [s@macro-outbound-callerid:10] Set("SIP/1000-091f5490", "EMERGENCYCID=<73614>") in new stack -- Executing [s@macro-outbound-callerid:11] Set("SIP/1000-091f5490", "TRUNKOUTCID=<73614>") in new stack -- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/1000-091f5490", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,16) -- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/1000-091f5490", "0?usercid") in new stack -- Executing [s@macro-outbound-callerid:17] Set("SIP/1000-091f5490", "CALLERID(all)=<73614>") in new stack -- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/1000-091f5490", "1?report") in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing [s@macro-outbound-callerid:22] NoOp("SIP/1000-091f5490", "CallerID set to "" <73614>") in new stack -- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/1000-091f5490", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,17) -- Executing [s@macro-dialout-trunk:17] AGI("SIP/1000-091f5490", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing [s@macro-dialout-trunk:18] Set("SIP/1000-091f5490", "OUTNUM=90222902") in new stack -- Executing [s@macro-dialout-trunk:19] Set("SIP/1000-091f5490", "custom=SIP/CBK") in new stack -- Executing [s@macro-dialout-trunk:20] GotoIf("SIP/1000-091f5490", "1?gocall") in new stack -- Goto (macro-dialout-trunk,s,22) -- Executing [s@macro-dialout-trunk:22] Macro("SIP/1000-091f5490", "dialout-trunk-predial-hook") in new stack -- Executing [s@macro-dialout-trunk:23] GotoIf("SIP/1000-091f5490", "0?bypass|1") in new stack -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/1000-091f5490", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:25] Dial("SIP/1000-091f5490", "SIP/CBK/90222902|300|TW") in new stack Audio is at 91.135.36.85 port 14664 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (NAT) to 88.87.41.169:5060: INVITE sip:90222902@sip.cbktele.com SIP/2.0 Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK220c9756;rport From: "73614" ;tag=as35d4b006 To: Contact: Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 05 Feb 2008 19:09:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 229 v=0 o=root 2570 2570 IN IP4 91.135.36.85 s=session c=IN IP4 91.135.36.85 t=0 0 m=audio 14664 RTP/AVP 0 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called CBK/90222902 trixbox1*CLI> <--- SIP read from 88.87.41.169:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK220c9756;received=91.135.36.85;rport=5060 From: "73614" ;tag=as35d4b006 To: ;tag=as342a6cbd Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85 CSeq: 102 INVITE User-Agent: CBK Telecom as Sip 0.9.50 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="260ff3e8" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (NAT) to 88.87.41.169:5060: ACK sip:90222902@sip.cbktele.com SIP/2.0 Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK220c9756;rport From: "73614" ;tag=as35d4b006 To: ;tag=as342a6cbd Contact: Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Audio is at 91.135.36.85 port 14664 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (NAT) to 88.87.41.169:5060: INVITE sip:90222902@sip.cbktele.com SIP/2.0 Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK728d2005;rport From: "73614" ;tag=as35d4b006 To: Contact: Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="73614", realm="asterisk", algorithm=MD5, uri="sip:90222902@sip.cbktele.com", nonce="260ff3e8", response="ca723957914e486213037a3b0a50ec5d", opaque="" Date: Tue, 05 Feb 2008 19:09:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 229 v=0 o=root 2570 2571 IN IP4 91.135.36.85 s=session c=IN IP4 91.135.36.85 t=0 0 m=audio 14664 RTP/AVP 0 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- trixbox1*CLI> <--- SIP read from 88.87.41.169:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK728d2005;received=91.135.36.85;rport=5060 From: "73614" ;tag=as35d4b006 To: Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85 CSeq: 103 INVITE User-Agent: CBK Telecom as Sip 0.9.50 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 88.87.41.169:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK728d2005;received=91.135.36.85;rport=5060 From: "73614" ;tag=as35d4b006 To: ;tag=as19604792 Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85 CSeq: 103 INVITE User-Agent: CBK Telecom as Sip 0.9.50 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 207 v=0 o=root 22909 22909 IN IP4 88.87.41.169 s=session c=IN IP4 88.87.41.169 t=0 0 m=audio 16392 RTP/AVP 8 3 0 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - <-------------> --- (11 headers 10 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 0 Peer audio RTP is at port 88.87.41.169:16392 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 88.87.41.169:16392 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 88.87.41.169, port 5060 Transmitting (NAT) to 88.87.41.169:5060: ACK sip:90222902@88.87.41.169 SIP/2.0 Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK55028cab;rport From: "73614" ;tag=as35d4b006 To: ;tag=as19604792 Contact: Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/CBK-09176620 answered SIP/1000-091f5490 Scheduling destruction of SIP dialog '1f914f3076a49fad32815ea9394f33dd@91.135.36.85' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 88.87.41.169, port 5060 Reliably Transmitting (NAT) to 88.87.41.169:5060: BYE sip:90222902@88.87.41.169 SIP/2.0 Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK632720cf;rport From: "73614" ;tag=as35d4b006 To: ;tag=as19604792 Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="73614", realm="asterisk", algorithm=MD5, uri="sip:90222902@88.87.41.169", nonce="260ff3e8", response="10fdbd78ec963af3ec24c29858014275", opaque="" Content-Length: 0 --- == Spawn extension (macro-dialout-trunk, s, 25) exited non-zero on 'SIP/1000-091f5490' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 25) exited non-zero on 'SIP/1000-091f5490' -- Executing [h@macro-dialout-trunk:1] Macro("SIP/1000-091f5490", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1000-091f5490", "w") in new stack trixbox1*CLI> <--- SIP read from 88.87.41.169:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 91.135.36.85:5060;branch=z9hG4bK632720cf;received=91.135.36.85;rport=5060 From: "73614" ;tag=as35d4b006 To: ;tag=as19604792 Call-ID: 1f914f3076a49fad32815ea9394f33dd@91.135.36.85 CSeq: 104 BYE User-Agent: CBK Telecom as Sip 0.9.50 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '1f914f3076a49fad32815ea9394f33dd@91.135.36.85' Method: INVITE -- Executing [s@macro-hangupcall:2] NoCDR("SIP/1000-091f5490", "") in new stack -- Executing [s@macro-hangupcall:3] GotoIf("SIP/1000-091f5490", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing [s@macro-hangupcall:6] GotoIf("SIP/1000-091f5490", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] GotoIf("SIP/1000-091f5490", "1?theend") in new stack -- Goto (macro-hangupcall,s,11) -- Executing [s@macro-hangupcall:11] Hangup("SIP/1000-091f5490", "") in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1000-091f5490' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1000-091f5490' Really destroying SIP dialog '644fe4dd2fee1498213c35055713507c@127.0.0.1' Method: REGISTER