rendered paste body
asterisk1*CLI>
<-- SIP read from 192.168.1.14:21680:
--- (0 headers 1 lines) ---
Destroying call 'Mjc4OWE5NjVhZjQ2NGQ0ZTY5NmNmZDY2NzZhMGRiYzY.'
Destroying call '4cf6aa554837e3b60a8105bd6f54a055@127.0.0.1'
asterisk1*CLI>
<-- SIP read from 212.98.65.15:32782:
INVITE sip:v2telin36938115@130.226.238.29 SIP/2.0
Via:SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-
From:<sip:26849929@212.98.65.15;user=phone>;tag=1949364555-1195663028222-
To:"36938115 36938115"<sip:36938115@commontrunk.v2tel.dk>
Call-ID:BW173708222211107-277357524@212.98.65.15
CSeq:831059968 INVITE
Contact:<sip:212.98.65.15:5060>
Supported:100rel
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept:multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards:10
Content-Type:application/sdp
Content-Disposition:session;handling=required
Content-Length:284
v=0
o=BroadWorks 117961895 1 IN IP4 212.98.65.50
s=-
c=IN IP4 212.98.65.50
t=0 0
m=audio 24900 RTP/AVP 8 18 0 101
c=IN IP4 212.98.65.50
a=fmtp:18 annexb=yes
a=fmtp:101 0-16
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
--- (14 headers 13 lines) ---
Using INVITE request as basis request - BW173708222211107-277357524@212.98.65.15
Sending to 212.98.65.15 : 5060 (NAT)
Found peer 'v2tel15'
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 212.98.65.50:24900
Found description format PCMA
Found description format G729
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for v2telin36938115 in from-trunk (domain 130.226.238.29)
list_route: hop: <sip:212.98.65.15:5060>
Transmitting (NAT) to 212.98.65.15:32782:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15
From: <sip:26849929@212.98.65.15;user=phone>;tag=1949364555-1195663028222-
To: "36938115 36938115"<sip:36938115@commontrunk.v2tel.dk>
Call-ID: BW173708222211107-277357524@212.98.65.15
CSeq: 831059968 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:v2telin36938115@130.226.238.29>
Content-Length: 0
---
-- Executing Set("SIP/36938115-09e71428", "__FROM_DID=v2telin36938115") in new stack
-- Executing Goto("SIP/36938115-09e71428", "ext-local|115|1") in new stack
-- Goto (ext-local,115,1)
-- Executing Macro("SIP/36938115-09e71428", "exten-vm|novm|115") in new stack
-- Executing Macro("SIP/36938115-09e71428", "user-callerid") in new stack
-- Executing NoOp("SIP/36938115-09e71428", "user-callerid: 26849929") in new stack
-- Executing GotoIf("SIP/36938115-09e71428", "0?report") in new stack
-- Executing GotoIf("SIP/36938115-09e71428", "0?start") in new stack
-- Executing Set("SIP/36938115-09e71428", "REALCALLERIDNUM=26849929") in new stack
-- Executing NoOp("SIP/36938115-09e71428", "REALCALLERIDNUM is 26849929") in new stack
-- Executing Set("SIP/36938115-09e71428", "AMPUSER=") in new stack
-- Executing Set("SIP/36938115-09e71428", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/36938115-09e71428", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/36938115-09e71428", "TTL: ARG1: novm") in new stack
-- Executing GotoIf("SIP/36938115-09e71428", "0?continue") in new stack
-- Executing Set("SIP/36938115-09e71428", "__TTL=64") in new stack
-- Executing GotoIf("SIP/36938115-09e71428", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/36938115-09e71428", "Using CallerID "" <26849929>") in new stack
-- Executing Set("SIP/36938115-09e71428", "FROMCONTEXT=exten-vm") in new stack
-- Executing Set("SIP/36938115-09e71428", "VMBOX=novm") in new stack
-- Executing Set("SIP/36938115-09e71428", "EXTTOCALL=115") in new stack
-- Executing Set("SIP/36938115-09e71428", "CFUEXT=") in new stack
-- Executing Set("SIP/36938115-09e71428", "CFBEXT=") in new stack
-- Executing Set("SIP/36938115-09e71428", "RT=") in new stack
-- Executing Macro("SIP/36938115-09e71428", "record-enable|115|IN") in new stack
-- Executing GotoIf("SIP/36938115-09e71428", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/36938115-09e71428", "recordingcheck|20071121-173708|1195663028.84") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20071121-173708|1195663028.84: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/36938115-09e71428", "No recording needed") in new stack
-- Executing Macro("SIP/36938115-09e71428", "dial||tr|115") in new stack
-- Executing DeadAGI("SIP/36938115-09e71428", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
asterisk1*CLI>
<-- SIP read from 212.98.65.15:32782:
INVITE sip:v2telin36938115@130.226.238.29 SIP/2.0
Via:SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-
From:<sip:26849929@212.98.65.15;user=phone>;tag=1949364555-1195663028222-
To:"36938115 36938115"<sip:36938115@commontrunk.v2tel.dk>
Call-ID:BW173708222211107-277357524@212.98.65.15
CSeq:831059968 INVITE
Contact:<sip:212.98.65.15:5060>
Supported:100rel
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept:multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards:10
Content-Type:application/sdp
Content-Disposition:session;handling=required
Content-Length:284
v=0
o=BroadWorks 117961895 1 IN IP4 212.98.65.50
s=-
c=IN IP4 212.98.65.50
t=0 0
m=audio 24900 RTP/AVP 8 18 0 101
c=IN IP4 212.98.65.50
a=fmtp:18 annexb=yes
a=fmtp:101 0-16
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
--- (14 headers 13 lines) ---
Ignoring this INVITE request
Transmitting (NAT) to 212.98.65.15:32782:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15
From: <sip:26849929@212.98.65.15;user=phone>;tag=1949364555-1195663028222-
To: "36938115 36938115"<sip:36938115@commontrunk.v2tel.dk>
Call-ID: BW173708222211107-277357524@212.98.65.15
CSeq: 831059968 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:v2telin36938115@130.226.238.29>
Content-Length: 0
---
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: priority is 1
dialparties.agi: Caller ID name is 'unknown' number is '26849929'
dialparties.agi: Methodology of ring is 'none'
> dialparties.agi: USE_CONFIRMATION: 'FALSE'
> dialparties.agi: RINGGROUP_INDEX: ''
-- dialparties.agi: Added extension 115 to extension map
-- dialparties.agi: Extension 115 cf is disabled
-- dialparties.agi: Extension 115 do not disturb is disabled
> dialparties.agi: extnum: 115
> dialparties.agi: exthascw: 1
> dialparties.agi: exthascfb: 0
> dialparties.agi: extcfb:
> dialparties.agi: exthascfu: 0
> dialparties.agi: extcfu:
-- dialparties.agi: dbset CALLTRACE/115 to 26849929
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/36938115-09e71428", "SIP/115||tr") in new stack
We're at 192.168.1.111 port 10514
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (NAT) to 192.168.1.13:17454:
INVITE sip:115@192.168.1.13:17454;rinstance=a0b7bdffdf6f5640 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK43a02646;rport
From: "26849929" <sip:26849929@192.168.1.111>;tag=as159d06dc
To: <sip:115@192.168.1.13:17454;rinstance=a0b7bdffdf6f5640>
Contact: <sip:26849929@192.168.1.111>
Call-ID: 44f7628f094a42de64b89e5714b000f6@192.168.1.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 21 Nov 2007 16:37:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 2523 2523 IN IP4 192.168.1.111
s=session
c=IN IP4 192.168.1.111
t=0 0
m=audio 10514 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 115
Transmitting (NAT) to 212.98.65.15:32782:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15
From: <sip:26849929@212.98.65.15;user=phone>;tag=1949364555-1195663028222-
To: "36938115 36938115"<sip:36938115@commontrunk.v2tel.dk>;tag=as55b58f2d
Call-ID: BW173708222211107-277357524@212.98.65.15
CSeq: 831059968 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:v2telin36938115@130.226.238.29>
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.13:17454:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK43a02646;rport=5060
Contact: <sip:115@192.168.1.13:17454;rinstance=a0b7bdffdf6f5640>
To: <sip:115@192.168.1.13:17454;rinstance=a0b7bdffdf6f5640>;tag=cf7ab53d
From: "26849929"<sip:26849929@192.168.1.111>;tag=as159d06dc
Call-ID: 44f7628f094a42de64b89e5714b000f6@192.168.1.111
CSeq: 102 INVITE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
--- (9 headers 0 lines) ---
-- SIP/115-09e769f0 is ringing
asterisk1*CLI>
<-- SIP read from 212.98.65.15:32782:
INVITE sip:v2telin36938115@130.226.238.29 SIP/2.0
Via:SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-
From:<sip:26849929@212.98.65.15;user=phone>;tag=1949364555-1195663028222-
To:"36938115 36938115"<sip:36938115@commontrunk.v2tel.dk>
Call-ID:BW173708222211107-277357524@212.98.65.15
CSeq:831059968 INVITE
Contact:<sip:212.98.65.15:5060>
Supported:100rel
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept:multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards:10
Content-Type:application/sdp
Content-Disposition:session;handling=required
Content-Length:284
v=0
o=BroadWorks 117961895 1 IN IP4 212.98.65.50
s=-
c=IN IP4 212.98.65.50
t=0 0
m=audio 24900 RTP/AVP 8 18 0 101
c=IN IP4 212.98.65.50
a=fmtp:18 annexb=yes
a=fmtp:101 0-16
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
--- (14 headers 13 lines) ---
Ignoring this INVITE request
Transmitting (NAT) to 212.98.65.15:32782:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15
From: <sip:26849929@212.98.65.15;user=phone>;tag=1949364555-1195663028222-
To: "36938115 36938115"<sip:36938115@commontrunk.v2tel.dk>
Call-ID: BW173708222211107-277357524@212.98.65.15
CSeq: 831059968 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:v2telin36938115@130.226.238.29>
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.5:52136:
--- (0 headers 1 lines) ---
asterisk1*CLI>
<-- SIP read from 192.168.1.3:15436:
--- (0 headers 1 lines) ---
asterisk1*CLI>
<-- SIP read from 212.98.65.15:32782:
INVITE sip:v2telin36938115@130.226.238.29 SIP/2.0
Via:SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-
From:<sip:26849929@212.98.65.15;user=phone>;tag=1949364555-1195663028222-
To:"36938115 36938115"<sip:36938115@commontrunk.v2tel.dk>
Call-ID:BW173708222211107-277357524@212.98.65.15
CSeq:831059968 INVITE
Contact:<sip:212.98.65.15:5060>
Supported:100rel
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept:multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards:10
Content-Type:application/sdp
Content-Disposition:session;handling=required
Content-Length:284
v=0
o=BroadWorks 117961895 1 IN IP4 212.98.65.50
s=-
c=IN IP4 212.98.65.50
t=0 0
m=audio 24900 RTP/AVP 8 18 0 101
c=IN IP4 212.98.65.50
a=fmtp:18 annexb=yes
a=fmtp:101 0-16
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
--- (14 headers 13 lines) ---
Ignoring this INVITE request
Transmitting (NAT) to 212.98.65.15:32782:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15
From: <sip:26849929@212.98.65.15;user=phone>;tag=1949364555-1195663028222-
To: "36938115 36938115"<sip:36938115@commontrunk.v2tel.dk>
Call-ID: BW173708222211107-277357524@212.98.65.15
CSeq: 831059968 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:v2telin36938115@130.226.238.29>
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.13:17454:
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK43a02646;rport=5060
To: <sip:115@192.168.1.13:17454;rinstance=a0b7bdffdf6f5640>;tag=cf7ab53d
From: "26849929"<sip:26849929@192.168.1.111>;tag=as159d06dc
Call-ID: 44f7628f094a42de64b89e5714b000f6@192.168.1.111
CSeq: 102 INVITE
User-Agent: X-Lite release 1011s stamp 41150
Warning: 399 devnull "User reject"
Content-Length: 0
--- (9 headers 0 lines) ---
-- Got SIP response 480 "Temporarily Unavailable" back from 192.168.1.13
Transmitting (NAT) to 192.168.1.13:17454:
ACK sip:115@192.168.1.13:17454;rinstance=a0b7bdffdf6f5640 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK43a02646;rport
From: "26849929" <sip:26849929@192.168.1.111>;tag=as159d06dc
To: <sip:115@192.168.1.13:17454;rinstance=a0b7bdffdf6f5640>;tag=cf7ab53d
Contact: <sip:26849929@192.168.1.111>
Call-ID: 44f7628f094a42de64b89e5714b000f6@192.168.1.111
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/115-09e769f0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Set("SIP/36938115-09e71428", "SV_DIALSTATUS=CONGESTION") in new stack
-- Executing GosubIf("SIP/36938115-09e71428", "0?docfu|1") in new stack
-- Executing GosubIf("SIP/36938115-09e71428", "0?docfb|1") in new stack
-- Executing Set("SIP/36938115-09e71428", "DIALSTATUS=CONGESTION") in new stack
-- Executing NoOp("SIP/36938115-09e71428", "Voicemail is novm") in new stack
-- Executing GotoIf("SIP/36938115-09e71428", "1?s-CONGESTION|1") in new stack
-- Goto (macro-exten-vm,s-CONGESTION,1)
-- Executing PlayTones("SIP/36938115-09e71428", "congestion") in new stack
We're at 130.226.238.29 port 14872
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 212.98.65.15:32782:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15
From: <sip:26849929@212.98.65.15;user=phone>;tag=1949364555-1195663028222-
To: "36938115 36938115"<sip:36938115@commontrunk.v2tel.dk>;tag=as55b58f2d
Call-ID: BW173708222211107-277357524@212.98.65.15
CSeq: 831059968 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:v2telin36938115@130.226.238.29>
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 2523 2523 IN IP4 130.226.238.29
s=session
c=IN IP4 130.226.238.29
t=0 0
m=audio 14872 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Executing Congestion("SIP/36938115-09e71428", "10") in new stack
Transmitting (NAT) to 212.98.65.15:32782:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 212.98.65.15;branch=z9hG4bK-BroadWorks.212.98.65.15-130.226.238.29V5060-0-831059968-1949364555-1195663028222-;received=212.98.65.15
From: <sip:26849929@212.98.65.15;user=phone>;tag=1949364555-1195663028222-
To: "36938115 36938115"<sip:36938115@commontrunk.v2tel.dk>;tag=as55b58f2d
Call-ID: BW173708222211107-277357524@212.98.65.15
CSeq: 831059968 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:v2telin36938115@130.226.238.29>
Content-Length: 0
X-Asterisk-HangupCause: User alerting, no answer
---
== Spawn extension (macro-exten-vm, s-CONGESTION, 2) exited non-zero on 'SIP/36938115-09e71428' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-CONGESTION, 2) exited non-zero on 'SIP/36938115-09e71428'
Destroying call '44f7628f094a42de64b89e5714b000f6@192.168.1.111'
Destroying call 'BW173708222211107-277357524@212.98.65.15'
asterisk1*CLI>
<-- SIP read from 192.168.1.2:36142:
--- (0 headers 1 lines) ---
asterisk1*CLI>
<-- SIP read from 192.168.1.13:17454:
--- (0 headers 1 lines) ---
asterisk1*CLI>