rendered paste body -- 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" <sip:sip1@24.45.38.133>;tag=as11e840ac
To: <sip:2261697@192.168.20.33:12>
Contact: <sip:sip1@24.45.38.133>
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: <sip:1001@192.168.20.33:12>
set_destination: Parsing <sip:1001@192.168.20.33:12> 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" <sip:sip1@24.45.38.133>;tag=as11e840ac
To: <sip:2261697@192.168.20.33:12>;tag=50a93aa0
Contact: <sip:sip1@24.45.38.133>
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 <sip:1001@192.168.20.33:12> 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" <sip:sip1@24.45.38.133>;tag=as11e840ac
To: <sip:2261697@192.168.20.33:12>;tag=50a93aa0
Contact: <sip:sip1@24.45.38.133>
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: <sip:2261697@192.168.20.33:12>;tag=50a93aa0
To: "6000" <sip:sip1@24.45.38.133>;tag=as11e840ac
Call-ID: 4dd431ad76c88e2759030f9f2392cc65@24.45.38.133
Date: Wed, 11 Jul 2007 13:25:15 GMT
Contact: <sip:1001@192.168.20.33:12>
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: <sip:2261697@192.168.20.33:12>;tag=50a93aa0
To: "6000" <sip:sip1@24.45.38.133>;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: <sip:sip1@24.45.38.133>
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'