=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2006.06.24 20:05:40 =~=~=~=~=~=~=~=~=~=~=~= -- Executing Macro("SIP/20001-16b3", "dialout-trunk|2|7279561||") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "DIAL_TRUNK=2") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "DIAL_NUMBER=7279561") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "ROUTE_PASSWD=") in new stack asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "1?noauth") in new stack asterisk1*CLI> -- Goto (macro-dialout-trunk,s,6) asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "GROUP()=OUT_2") in new stack asterisk1*CLI> -- Executing Macro("SIP/20001-16b3", "user-callerid") in new stack asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "0?report") in new stack asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "0?start") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "REALCALLERIDNUM=20001") in new stack asterisk1*CLI> -- Executing NoOp("SIP/20001-16b3", "REALCALLERIDNUM is 20001") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "AMPUSER=20001") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "AMPUSERCIDNAME=20001") in new stack asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "0?report") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "CALLERID(all)=20001 <20001>") in new stack asterisk1*CLI> -- Executing NoOp("SIP/20001-16b3", "Using CallerID "20001" <20001>") in new stack asterisk1*CLI> -- Executing Macro("SIP/20001-16b3", "record-enable|20001|OUT") in new stack asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "0 > 0?2:4") in new stack asterisk1*CLI> -- Goto (macro-record-enable,s,4) asterisk1*CLI> -- Executing AGI("SIP/20001-16b3", "recordingcheck|20060624-190603|1151150763.55") in new stack asterisk1*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck asterisk1*CLI> recordingcheck|20060624-190603|1151150763.55: Outbound recording not enabled asterisk1*CLI> -- AGI Script recordingcheck completed, returning 0 asterisk1*CLI> -- Executing NoOp("SIP/20001-16b3", "No recording needed") in new stack asterisk1*CLI> -- Executing Macro("SIP/20001-16b3", "outbound-callerid|2") in new stack asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "1?start") in new stack asterisk1*CLI> -- Goto (macro-outbound-callerid,s,3) asterisk1*CLI> -- Executing NoOp("SIP/20001-16b3", "REALCALLERIDNUM is 20001") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "USEROUTCID=") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "EMERGENCYCID=") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "TRUNKOUTCID=1656222") in new stack asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "1?trunkcid") in new stack asterisk1*CLI> -- Goto (macro-outbound-callerid,s,11) asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "0?usercid") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "CALLERID(all)=1656222") in new stack asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "1?report") in new stack asterisk1*CLI> -- Goto (macro-outbound-callerid,s,15) asterisk1*CLI> -- Executing NoOp("SIP/20001-16b3", "CallerID set to "" <1656222>") in new stack asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "0?nomax") in new stack asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "0?chanfull") in new stack asterisk1*CLI> -- Executing AGI("SIP/20001-16b3", "fixlocalprefix") in new stack asterisk1*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix asterisk1*CLI> fixlocalprefix: Added prefix. New number: 16562227279561 asterisk1*CLI> -- AGI Script fixlocalprefix completed, returning 0 asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "OUTNUM=16562227279561") in new stack asterisk1*CLI> -- Executing Set("SIP/20001-16b3", "custom=SIP/SIP_BD1") in new stack asterisk1*CLI> -- Executing GotoIf("SIP/20001-16b3", "0?customtrunk") in new stack asterisk1*CLI> -- Executing Dial("SIP/20001-16b3", "SIP/SIP_BD1/16562227279561|120|r") in new stack asterisk1*CLI> We're at 192.168.0.1 port 17030 asterisk1*CLI> Adding codec 0x1 (g723) to SDP asterisk1*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk1*CLI> 13 headers, 10 lines asterisk1*CLI> Reliably Transmitting (no NAT) to 192.168.0.254:5060: INVITE sip:16562227279561@192.168.0.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0acb1cb7;rport From: "1656222" ;tag=as5701409f To: Contact: Call-ID: 63efe37001992e282155433d048f3a10@192.168.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 24 Jun 2006 12:06:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 2646 2646 IN IP4 192.168.0.1 s=session c=IN IP4 192.168.0.1 t=0 0 m=audio 17030 RTP/AVP 4 101 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk1*CLI> -- Called SIP_BD1/16562227279561 asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 100 Trying Call-ID: 63efe37001992e282155433d048f3a10@192.168.0.1 CSeq: 102 INVITE From: "1656222";tag=as5701409f To: ;tag=c0a800fe-4bc User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0acb1cb7;rport Quintum: 0b023232 --- (8 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 183 Session Progress Call-ID: 63efe37001992e282155433d048f3a10@192.168.0.1 Content-Length: 160 Content-Type: application/sdp CSeq: 102 INVITE From: "1656222";tag=as5701409f To: ;tag=c0a800fe-4bc User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0acb1cb7;rport Quintum: 070e00000003008f6506001e03808081 v=0 o=Quintum 8 2646 IN IP4 192.168.0.254 s=VoipCall c=IN IP4 192.168.0.254 t=0 0 m=audio 10240 RTP/AVP 4 c=IN IP4 192.168.0.254 a=rtpmap:4 g723/8000/1 --- (10 headers 8 lines)--- asterisk1*CLI> Found RTP audio format 4 Peer audio RTP is at port 192.168.0.254:10240 Found description format g723 Capabilities: us - 0x1 (g723), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) -- SIP/SIP_BD1-113a is making progress passing it to SIP/20001-16b3 asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 180 Ringing Call-ID: 63efe37001992e282155433d048f3a10@192.168.0.1 Content-Length: 160 Content-Type: application/sdp CSeq: 102 INVITE From: "1656222";tag=as5701409f To: ;tag=c0a800fe-4bc User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0acb1cb7;rport v=0 o=Quintum 9 2646 IN IP4 192.168.0.254 s=VoipCall c=IN IP4 192.168.0.254 t=0 0 m=audio 10240 RTP/AVP 4 c=IN IP4 192.168.0.254 a=rtpmap:4 g723/8000/1 --- (9 headers 8 lines)--- Found RTP audio format 4 Peer audio RTP is at port 192.168.0.254:10240 Found description format g723 Capabilities: us - 0x1 (g723), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) -- SIP/SIP_BD1-113a is ringing -- SIP/SIP_BD1-113a is making progress passing it to SIP/20001-16b3 asterisk1*CLI> 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.0.254:5060: OPTIONS sip:192.168.0.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK203989a9;rport From: "Unknown" ;tag=as22d29532 To: Contact: Call-ID: 7f8562936a8644bf719d23ec7e192f47@192.168.0.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 24 Jun 2006 12:06:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 200 OK Call-ID: 7f8562936a8644bf719d23ec7e192f47@192.168.0.1 CSeq: 102 OPTIONS From: "Unknown";tag=as22d29532 To: Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK203989a9;rport --- (6 headers 0 lines)--- Destroying call '7f8562936a8644bf719d23ec7e192f47@192.168.0.1' asterisk1*CLI> Reliably Transmitting (no NAT) to 192.168.0.254:5060: CANCEL sip:16562227279561@192.168.0.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0acb1cb7;rport From: "1656222" ;tag=as5701409f To: Contact: Call-ID: 63efe37001992e282155433d048f3a10@192.168.0.1 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '63efe37001992e282155433d048f3a10@192.168.0.1' in 32000 ms == Spawn extension (macro-dialout-trunk, s, 16) exited non-zero on 'SIP/20001-16b3' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 16) exited non-zero on 'SIP/20001-16b3' asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 200 OK Call-ID: 63efe37001992e282155433d048f3a10@192.168.0.1 CSeq: 102 CANCEL From: "1656222";tag=as5701409f To: Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0acb1cb7;rport --- (6 headers 0 lines)--- asterisk1*CLI>