-- Executing Set("SIP/222-b710f1b0", "REALCALLERIDNUM=222") in new stack -- Executing NoOp("SIP/222-b710f1b0", "TTL: ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/222-b710f1b0", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/222-b710f1b0", "Using CallerID "222" <222>") in new stack -- Executing Macro("SIP/222-b710f1b0", "record-enable|222|OUT") in new stack -- Executing GotoIf("SIP/222-b710f1b0", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/222-b710f1b0", "recordingcheck|20070417-200637|pbx-4063-1176833197.4") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070417-200637|pbx-4063-1176833197.4: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/222-b710f1b0", "No recording needed") in new stack -- Executing GotoIf("SIP/222-b710f1b0", "0?skipoutcid") in new stack -- Executing Set("SIP/222-b710f1b0", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing Macro("SIP/222-b710f1b0", "outbound-callerid|2") in new stack -- Executing GotoIf("SIP/222-b710f1b0", "1?start") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp("SIP/222-b710f1b0", "REALCALLERIDNUM is 222") in new stack -- Executing GotoIf("SIP/222-b710f1b0", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,9) -- Executing Set("SIP/222-b710f1b0", "USEROUTCID=") in new stack -- Executing Set("SIP/222-b710f1b0", "EMERGENCYCID=") in new stack -- Executing Set("SIP/222-b710f1b0", "TRUNKOUTCID=") in new stack -- Executing GotoIf("SIP/222-b710f1b0", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,16) -- Executing GotoIf("SIP/222-b710f1b0", "1?usercid") in new stack -- Goto (macro-outbound-callerid,s,18) -- Executing GotoIf("SIP/222-b710f1b0", "1?report") in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing NoOp("SIP/222-b710f1b0", "CallerID set to "222" <222>") in new stack -- Executing GotoIf("SIP/222-b710f1b0", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,16) -- Executing DeadAGI("SIP/222-b710f1b0", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/222-b710f1b0", "OUTNUM=27844045791") in new stack -- Executing Set("SIP/222-b710f1b0", "custom=SIP/GOVOIP") in new stack -- Executing GotoIf("SIP/222-b710f1b0", "0?customtrunk") in new stack -- Executing Dial("SIP/222-b710f1b0", "SIP/GOVOIP/27844045791|300|") in new stack We're at 41.242.87.135 port 19716 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 13 lines Reliably Transmitting (NAT) to 41.204.198.76:5060: INVITE sip:27844045791@sip.voipstream.co.za SIP/2.0 Via: SIP/2.0/UDP 41.242.87.135:5060;branch=z9hG4bK5e6d8989;rport From: "222" ;tag=as59f3b86b To: Contact: Call-ID: 1147c5ae6f8d8eb566b5969b33f9ac87@41.242.87.135 CSeq: 102 INVITE User-Agent: Asterisk PBX ax-Forwards: 70 Date: Tue, 17 Apr 2007 18:06:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 287 v=0 o=root 4063 4063 IN IP4 41.242.87.135 s=session c=IN IP4 41.242.87.135 t=0 0 m=audio 19716 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called GOVOIP/27844045791 Retransmitting #1 (NAT) to 41.204.198.76:5060: INVITE sip:27844045791@sip.voipstream.co.za SIP/2.0 Via: SIP/2.0/UDP 41.242.87.135:5060;branch=z9hG4bK5e6d8989;rport From: "222" ;tag=as59f3b86b To: Contact: Call-ID: 1147c5ae6f8d8eb566b5969b33f9ac87@41.242.87.135 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 17 Apr 2007 18:06:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 287 v=0 o=root 4063 4063 IN IP4 41.242.87.135 s=session c=IN IP4 41.242.87.135 t=0 0 m=audio 19716 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (NAT) to 41.204.198.76:5060: INVITE sip:27844045791@sip.voipstream.co.za SIP/2.0 Via: SIP/2.0/UDP 41.242.87.135:5060;branch=z9hG4bK5e6d8989;rport From: "222" ;tag=as59f3b86b To: Contact: Call-ID: 1147c5ae6f8d8eb566b5969b33f9ac87@41.242.87.135 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 17 Apr 2007 18:06:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 287 v=0 o=root 4063 4063 IN IP4 41.242.87.135 s=session c=IN IP4 41.242.87.135 t=0 0 m=audio 19716 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (NAT) to 41.204.198.76:5060: INVITE sip:27844045791@sip.voipstream.co.za SIP/2.0 Via: SIP/2.0/UDP 41.242.87.135:5060;branch=z9hG4bK5e6d8989;rport From: "222" ;tag=as59f3b86b To: Contact: Call-ID: 1147c5ae6f8d8eb566b5969b33f9ac87@41.242.87.135 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 17 Apr 2007 18:06:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 287 v=0 o=root 4063 4063 IN IP4 41.242.87.135 s=session c=IN IP4 41.242.87.135 t=0 0 m=audio 19716 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #4 (NAT) to 41.204.198.76:5060: INVITE sip:27844045791@sip.voipstream.co.za SIP/2.0 Via: SIP/2.0/UDP 41.242.87.135:5060;branch=z9hG4bK5e6d8989;rport From: "222" ;tag=as59f3b86b To: Contact: Call-ID: 1147c5ae6f8d8eb566b5969b33f9ac87@41.242.87.135 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 17 Apr 2007 18:06:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 287 v=0 o=root 4063 4063 IN IP4 41.242.87.135 s=session c=IN IP4 41.242.87.135 t=0 0 m=audio 19716 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Destroying call '37a550006564bd900950424a000cd422@127.0.1.1' Retransmitting #5 (NAT) to 41.204.198.76:5060: INVITE sip:27844045791@sip.voipstream.co.za SIP/2.0 Via: SIP/2.0/UDP 41.242.87.135:5060;branch=z9hG4bK5e6d8989;rport From: "222" ;tag=as59f3b86b To: Contact: Call-ID: 1147c5ae6f8d8eb566b5969b33f9ac87@41.242.87.135 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 17 Apr 2007 18:06:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 287 v=0 o=root 4063 4063 IN IP4 41.242.87.135 s=session c=IN IP4 41.242.87.135 t=0 0 m=audio 19716 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Scheduling destruction of call '1147c5ae6f8d8eb566b5969b33f9ac87@41.242.87.135' in 32000 ms == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/222-b710f1b0' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/222-b710f1b0' -- Executing Macro("SIP/222-b710f1b0", "hangupcall") in new stack -- Executing ResetCDR("SIP/222-b710f1b0", "w") in new stack -- Executing NoCDR("SIP/222-b710f1b0", "") in new stack -- Executing GotoIf("SIP/222-b710f1b0", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf("SIP/222-b710f1b0", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing Wait("SIP/222-b710f1b0", "5") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/222-b710f1b0' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/222-b710f1b0'