rendered paste body -- 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" <sip:222@41.242.87.135>;tag=as59f3b86b
To: <sip:27844045791@sip.voipstream.co.za>
Contact: <sip:222@41.242.87.135>
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" <sip:222@41.242.87.135>;tag=as59f3b86b
To: <sip:27844045791@sip.voipstream.co.za>
Contact: <sip:222@41.242.87.135>
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" <sip:222@41.242.87.135>;tag=as59f3b86b
To: <sip:27844045791@sip.voipstream.co.za>
Contact: <sip:222@41.242.87.135>
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" <sip:222@41.242.87.135>;tag=as59f3b86b
To: <sip:27844045791@sip.voipstream.co.za>
Contact: <sip:222@41.242.87.135>
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" <sip:222@41.242.87.135>;tag=as59f3b86b
To: <sip:27844045791@sip.voipstream.co.za>
Contact: <sip:222@41.242.87.135>
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" <sip:222@41.242.87.135>;tag=as59f3b86b
To: <sip:27844045791@sip.voipstream.co.za>
Contact: <sip:222@41.242.87.135>
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'