rendered paste body=~=~=~=~=~=~=~=~=~=~=~= 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" <sip:1656222@192.168.0.1>;tag=as5701409f
To: <sip:16562227279561@192.168.0.254>
Contact: <sip:1656222@192.168.0.1>
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"<sip:1656222@192.168.0.1>;tag=as5701409f
To: <sip:16562227279561@192.168.0.254>;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"<sip:1656222@192.168.0.1>;tag=as5701409f
To: <sip:16562227279561@192.168.0.254>;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"<sip:1656222@192.168.0.1>;tag=as5701409f
To: <sip:16562227279561@192.168.0.254>;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" <sip:Unknown@192.168.0.1>;tag=as22d29532
To: <sip:192.168.0.254>
Contact: <sip:Unknown@192.168.0.1>
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"<sip:Unknown@192.168.0.1>;tag=as22d29532
To: <sip:192.168.0.254>
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" <sip:1656222@192.168.0.1>;tag=as5701409f
To: <sip:16562227279561@192.168.0.254>
Contact: <sip:1656222@192.168.0.1>
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"<sip:1656222@192.168.0.1>;tag=as5701409f
To: <sip:16562227279561@192.168.0.254>
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0acb1cb7;rport
--- (6 headers 0 lines)---
asterisk1*CLI>