-- Executing Macro("SIP/6000-b781a940", "dialout-trunk|6|2261697||") in new stack -- Executing Set("SIP/6000-b781a940", "DIAL_TRUNK=6") in new stack -- Executing Set("SIP/6000-b781a940", "_NODEST=") in new stack -- Executing Set("SIP/6000-b781a940", "DIAL_NUMBER=2261697") in new stack -- Executing Set("SIP/6000-b781a940", "ROUTE_PASSWD=") in new stack -- Executing Set("SIP/6000-b781a940", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing GotoIf("SIP/6000-b781a940", "1?noauth") in new stack -- Goto (macro-dialout-trunk,s,8) -- Executing Set("SIP/6000-b781a940", "GROUP()=OUT_6") in new stack -- Executing Macro("SIP/6000-b781a940", "user-callerid|SKIPTTL") in new stack -- Executing NoOp("SIP/6000-b781a940", "user-callerid: device 6000") in new stack -- Executing GotoIf("SIP/6000-b781a940", "0?report") in new stack -- Executing GotoIf("SIP/6000-b781a940", "0?start") in new stack -- Executing Set("SIP/6000-b781a940", "REALCALLERIDNUM=6000") in new stack -- Executing NoOp("SIP/6000-b781a940", "REALCALLERIDNUM is 6000") in new stack -- Executing Set("SIP/6000-b781a940", "AMPUSER=6000") in new stack -- Executing Set("SIP/6000-b781a940", "AMPUSERCIDNAME=6000") in new stack -- Executing GotoIf("SIP/6000-b781a940", "0?report") in new stack -- Executing Set("SIP/6000-b781a940", "CALLERID(all)=6000 <6000>") in new stack -- Executing Set("SIP/6000-b781a940", "REALCALLERIDNUM=6000") in new stack -- Executing NoOp("SIP/6000-b781a940", "TTL: ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/6000-b781a940", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/6000-b781a940", "Using CallerID "6000" <6000>") in new stack -- Executing Macro("SIP/6000-b781a940", "record-enable|6000|OUT") in new stack -- Executing GotoIf("SIP/6000-b781a940", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/6000-b781a940", "recordingcheck|20070711-132515|1184160315.20375") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070711-132515|1184160315.20375: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/6000-b781a940", "No recording needed") in new stack -- Executing GotoIf("SIP/6000-b781a940", "0?skipoutcid") in new stack -- Executing Set("SIP/6000-b781a940", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing Macro("SIP/6000-b781a940", "outbound-callerid|6") in new stack -- Executing GotoIf("SIP/6000-b781a940", "1?start") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp("SIP/6000-b781a940", "REALCALLERIDNUM is 6000") in new stack -- Executing GotoIf("SIP/6000-b781a940", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,9) -- Executing Set("SIP/6000-b781a940", "USEROUTCID=") in new stack -- Executing Set("SIP/6000-b781a940", "EMERGENCYCID=") in new stack -- Executing Set("SIP/6000-b781a940", "TRUNKOUTCID=") in new stack -- Executing GotoIf("SIP/6000-b781a940", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,16) -- Executing GotoIf("SIP/6000-b781a940", "1?usercid") in new stack -- Goto (macro-outbound-callerid,s,18) -- Executing GotoIf("SIP/6000-b781a940", "1?report") in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing NoOp("SIP/6000-b781a940", "CallerID set to "6000" <6000>") in new stack -- Executing GotoIf("SIP/6000-b781a940", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,16) -- Executing DeadAGI("SIP/6000-b781a940", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/6000-b781a940", "OUTNUM=2261697") in new stack -- Executing Set("SIP/6000-b781a940", "custom=SIP/GSM") in new stack -- Executing GotoIf("SIP/6000-b781a940", "0?customtrunk") in new stack -- Executing Dial("SIP/6000-b781a940", "SIP/GSM/2261697|300|") in new stack We're at 24.45.38.133 port 17712 Adding codec 0x100 (g729) to SDP 13 headers, 9 lines Reliably Transmitting (no NAT) to 192.168.20.33:12: INVITE sip:2261697@192.168.20.33:12 SIP/2.0 Via: SIP/2.0/UDP 24.45.38.133:5060;branch=z9hG4bK7e2881d0;rport From: "6000" ;tag=as11e840ac To: Contact: Call-ID: 4dd431ad76c88e2759030f9f2392cc65@24.45.38.133 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 11 Jul 2007 13:25:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 181 v=0 o=root 3207 3207 IN IP4 24.45.38.133 s=session c=IN IP4 24.45.38.133 t=0 0 m=audio 17712 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - --- -- Called GSM/2261697 -- SIP/GSM-09df9948 is ringing Found RTP audio format 18 Peer audio RTP is at port 192.168.20.33:60000 Found description format G729 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.20.33, port 12 Transmitting (no NAT) to 192.168.20.33:12: ACK sip:1001@192.168.20.33:12 SIP/2.0 Via: SIP/2.0/UDP 24.45.38.133:5060;branch=z9hG4bK512addba;rport From: "6000" ;tag=as11e840ac To: ;tag=50a93aa0 Contact: Call-ID: 4dd431ad76c88e2759030f9f2392cc65@24.45.38.133 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/GSM-09df9948 answered SIP/6000-b781a940 -- Attempting native bridge of SIP/6000-b781a940 and SIP/GSM-09df9948 Found RTP audio format 18 Peer audio RTP is at port 192.168.20.33:60000 Found description format G729 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.20.33, port 12 Transmitting (no NAT) to 192.168.20.33:12: ACK sip:1001@192.168.20.33:12 SIP/2.0 Via: SIP/2.0/UDP 24.45.38.133:5060;branch=z9hG4bK5eab0d8f;rport From: "6000" ;tag=as11e840ac To: ;tag=50a93aa0 Contact: Call-ID: 4dd431ad76c88e2759030f9f2392cc65@24.45.38.133 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.20.33:12: BYE sip:192.168.11.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.33:12;rport;branch=z9hG4bK8fa65ab980408e71ab7148d329c0d4eb From: ;tag=50a93aa0 To: "6000" ;tag=as11e840ac Call-ID: 4dd431ad76c88e2759030f9f2392cc65@24.45.38.133 Date: Wed, 11 Jul 2007 13:25:15 GMT Contact: CSeq: 801 BYE Max-Forwards: 70 User-Agent: MV-37x Content-Length: 0 --- (11 headers 0 lines) --- Sending to 192.168.20.33 : 12 (NAT) Transmitting (NAT) to 192.168.20.33:12: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.33:12;branch=z9hG4bK8fa65ab980408e71ab7148d329c0d4eb;received=192.168.20.33;rport=12 From: ;tag=50a93aa0 To: "6000" ;tag=as11e840ac Call-ID: 4dd431ad76c88e2759030f9f2392cc65@24.45.38.133 CSeq: 801 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/6000-b781a940' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/6000-b781a940' -- Executing Macro("SIP/6000-b781a940", "hangupcall") in new stack -- Executing ResetCDR("SIP/6000-b781a940", "w") in new stack -- Executing NoCDR("SIP/6000-b781a940", "") in new stack -- Executing GotoIf("SIP/6000-b781a940", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf("SIP/6000-b781a940", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing Wait("SIP/6000-b781a940", "5") in new stack Destroying call '4dd431ad76c88e2759030f9f2392cc65@24.45.38.133' -- Executing Hangup("SIP/6000-b781a940", "") in new stack == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/6000-b781a940' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/6000-b781a940'