rendered paste body# Example CDR Entry
| 2007-06-13 14:32:40 | "KAYAK.COM " <2038993102> | 2038993102 | 5132 | default | SIP/5060-b69042c0 | SIP/5132-08238878 | Dial | SIP/5132|18 | 91 | 86 | ANSWERED | 3 | | |
# extensions.conf (pertinent section)
[DID]
exten => _5[1-2]XX,hint,SIP/${EXTEN}
exten => _5[1-2]XX,1,Dial(SIP/${EXTEN},18)
exten => _5[1-2]XX,2,NoOp(${CALLERID(num)}---${CALLERID(name)})
exten => _5[1-2]XX,3,GotoIfTime(09:00-22:00,mon-sun,*,*?5)
exten => _5[1-2]XX,4,Goto(6)
exten => _5[1-2]XX,5,FollowMe(follow${EXTEN}|san)
exten => _5[1-2]XX,6,Voicemail(${EXTEN}@allstaff,u)
exten => _5[1-2]XX,7,Hangup()
exten => _5[1-2]XX,102,Voicemail(${EXTEN}@allstaff,b)
exten => _5[1-2]XX,103,Hangup()
# SIP Debugging
grosvenor*CLI> sip set debug
SIP Debugging enabled
grosvenor*CLI> sip set debug off
<--- SIP read from 10.10.10.5:5060 --->
INVITE sip:5201;npdi@192.168.1.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.5:5060;branch=z9hG4bK00k2db10e81gte4fu6c0.1
From: "Unavailable"<sip:2012147154@10.10.10.5:5060;isup-oli=62;user=phone;2649054plex=2649054plex-elchk8eg33lcb>;tag=127.0.0.1+1+be680005+5b4d5bcb
To: <sip:5201@192.168.1.2:5060;user=phone>
CSeq: 761307750 INVITE
Expires: 180
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, PRACK, INFO, UPDATE
Call-ID: 280C0958-1ACC97@10.254.1.7
-Asserted-Identity: "Unavailable"<sip:2012147154@10.254.1.7;user=phone>
Privacy: none
Max-Forwards: 56
Contact: "Unavailable"<sip:2012147154@10.10.10.5:5060;isup-oli=62;2649054plex=2649054plex-elchk8eg33lcb;transport=udp>
Content-Type: application/sdp
Content-Length: 327
v=0
o=- 3390753944 3390753944 IN IP4 10.10.10.5
s=-
c=IN IP4 10.10.10.5
t=0 0
m=audio 20008 RTP/AVP 18 0 8 98 101
a=ptime:20
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 clearmode/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -
<------------->
--- (14 headers 15 lines) ---
Sending to 10.10.10.5 : 5060 (no NAT)
Using INVITE request as basis request - 280C0958-1ACC97@10.254.1.7
Found peer 'paetec'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 10.10.10.5:20008
Found description format G729 for ID 18
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format clearmode for ID 98
Found description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.10.10.5:20008
Looking for 5201 in default (domain 192.168.1.2)
list_route: hop: <sip:2012147154@10.10.10.5:5060;isup-oli=62;2649054plex=2649054plex-elchk8eg33lcb;transport=udp>
<--- Transmitting (no NAT) to 10.10.10.5:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.5:5060;branch=z9hG4bK00k2db10e81gte4fu6c0.1;received=10.10.10.5
From: "Unavailable"<sip:2012147154@10.10.10.5:5060;isup-oli=62;user=phone;2649054plex=2649054plex-elchk8eg33lcb>;tag=127.0.0.1+1+be680005+5b4d5bcb
To: <sip:5201@192.168.1.2:5060;user=phone>
Call-ID: 280C0958-1ACC97@10.254.1.7
CSeq: 761307750 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5201@192.168.1.2>
Content-Length: 0
<------------>
Audio is at 192.168.0.205 port 11374
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.201:5060:
INVITE sip:5201@192.168.0.201:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK39a41850;rport
From: "Unavailable" <sip:5201@192.168.0.205>;tag=as36bd68c6
To: <sip:5201@192.168.0.201:5060;transport=udp>
Contact: <sip:5201@192.168.0.205>
Call-ID: 2d503e1c4a98fd3f776b447238e00600@192.168.0.205
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Jun 2007 20:05:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 22739 22739 IN IP4 192.168.0.205
s=session
c=IN IP4 192.168.0.205
t=0 0
m=audio 11374 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
grosvenor*CLI> sip set debug off
<--- SIP read from 192.168.0.201:49157 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK39a41850;rport
From: "Unavailable" <sip:5201@192.168.0.205>;tag=as36bd68c6
To: <sip:5201@192.168.0.201:5060;transport=udp>
Call-ID: 2d503e1c4a98fd3f776b447238e00600@192.168.0.205
Date: Wed, 13 Jun 2007 GMT
CSeq: 102 INVITE
Server: Cisco-CP7961G/8.0
Contact: <sip:5201@192.168.0.201:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Allow-Events: kpml,dialog
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
grosvenor*CLI> sip set debug off
<--- SIP read from 192.168.0.201:49157 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK39a41850;rport
From: "Unavailable" <sip:5201@192.168.0.205>;tag=as36bd68c6
To: <sip:5201@192.168.0.201:5060;transport=udp>;tag=0019e8f4b4d0181a883d0369-e4a7fe5e
Call-ID: 2d503e1c4a98fd3f776b447238e00600@192.168.0.205
Date: Wed, 13 Jun 2007 GMT
CSeq: 102 INVITE
Server: Cisco-CP7961G/8.0
Contact: <sip:5201@192.168.0.201:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "Eric Lubow" <sip:5201@192.168.0.205>;party=called;id-type=subscriber;privacy=off;screen=yes
Allow-Events: kpml,dialog
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
grosvenor*CLI> sip set debug off
<--- Transmitting (no NAT) to 10.10.10.5:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.5:5060;branch=z9hG4bK00k2db10e81gte4fu6c0.1;received=10.10.10.5
From: "Unavailable"<sip:2012147154@10.10.10.5:5060;isup-oli=62;user=phone;2649054plex=2649054plex-elchk8eg33lcb>;tag=127.0.0.1+1+be680005+5b4d5bcb
To: <sip:5201@192.168.1.2:5060;user=phone>;tag=as40e6688f
Call-ID: 280C0958-1ACC97@10.254.1.7
CSeq: 761307750 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5201@192.168.1.2>
Content-Length: 0
<------------>
grosvenor*CLI> sip set debug off
<--- SIP read from 192.168.0.201:49157 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK39a41850;rport
From: "Unavailable" <sip:5201@192.168.0.205>;tag=as36bd68c6
To: <sip:5201@192.168.0.201:5060;transport=udp>;tag=0019e8f4b4d0181a883d0369-e4a7fe5e
Call-ID: 2d503e1c4a98fd3f776b447238e00600@192.168.0.205
Date: Wed, 13 Jun 2007 GMT
CSeq: 102 INVITE
Server: Cisco-CP7961G/8.0
Contact: <sip:5201@192.168.0.201:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "Eric Lubow" <sip:5201@192.168.0.205>;party=called;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 206
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 9435 0 IN IP4 192.168.0.201
s=SIP Call
t=0 0
m=audio 19226 RTP/AVP 0 101
c=IN IP4 192.168.0.201
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
-------------> sip set debug off
--- (16 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.201:19226
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.201:19226
list_route: hop: <sip:5201@192.168.0.201:5060;transport=udp>
set_destination: Parsing <sip:5201@192.168.0.201:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.0.201, port 5060
Transmitting (no NAT) to 192.168.0.201:5060:
ACK sip:5201@192.168.0.201:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.205:5060;branch=z9hG4bK635c94dd;rport
From: "Unavailable" <sip:5201@192.168.0.205>;tag=as36bd68c6
To: <sip:5201@192.168.0.201:5060;transport=udp>;tag=0019e8f4b4d0181a883d0369-e4a7fe5e
Contact: <sip:5201@192.168.0.205>
Call-ID: 2d503e1c4a98fd3f776b447238e00600@192.168.0.205
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Audio is at 192.168.1.2 port 12610
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
grosvenor*CLI> sip set debug off
<--- Reliably Transmitting (no NAT) to 10.10.10.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.5:5060;branch=z9hG4bK00k2db10e81gte4fu6c0.1;received=10.10.10.5
From: "Unavailable"<sip:2012147154@10.10.10.5:5060;isup-oli=62;user=phone;2649054plex=2649054plex-elchk8eg33lcb>;tag=127.0.0.1+1+be680005+5b4d5bcb
To: <sip:5201@192.168.1.2:5060;user=phone>;tag=as40e6688f
Call-ID: 280C0958-1ACC97@10.254.1.7
CSeq: 761307750 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5201@192.168.1.2>
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 22739 22739 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
t=0 0
m=audio 12610 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
grosvenor*CLI> sip set debug off
<--- SIP read from 10.10.10.5:5060 --->
ACK sip:5201@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.5:5060;branch=z9hG4bK00k2dc109oph2eo420g0.1
Call-ID: 280C0958-1ACC97@10.254.1.7
From: "Unavailable"<sip:2012147154@10.10.10.5:5060;isup-oli=62;user=phone;2649054plex=2649054plex-elchk8eg33lcb>;tag=127.0.0.1+1+be680005+5b4d5bcb
To: <sip:5201@192.168.1.2:5060;user=phone>;tag=as40e6688f
CSeq: 761307750 ACK
Contact: "Unavailable"<sip:2012147154@10.10.10.5:5060;isup-oli=62;2649054plex=2649054plex-elchk8eg33lcb;transport=udp>
Content-Length: 0
Max-Forwards: 56
<------------->
--- (9 headers 0 lines) ---
grosvenor*CLI> sip set debug off
<--- SIP read from 192.168.0.201:49157 --->
BYE sip:5201@192.168.0.205:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bKe719b825
From: <sip:5201@192.168.0.201:5060;transport=udp>;tag=0019e8f4b4d0181a883d0369-e4a7fe5e
To: "Unavailable" <sip:5201@192.168.0.205>;tag=as36bd68c6
Call-ID: 2d503e1c4a98fd3f776b447238e00600@192.168.0.205
Max-Forwards: 70
Date: Wed, 13 Jun 2007 GMT
CSeq: 101 BYE
User-Agent: Cisco-CP7961G/8.0
Content-Length: 0
RTP-RxStat: Dur=4,Pkt=191,Oct=32852,LatePkt=0,LostPkt=0,AvgJit=0
RTP-TxStat: Dur=4,Pkt=194,Oct=33368
<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.201 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.0.201:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.201:5060;branch=z9hG4bKe719b825;received=192.168.0.201
From: <sip:5201@192.168.0.201:5060;transport=udp>;tag=0019e8f4b4d0181a883d0369-e4a7fe5e
To: "Unavailable" <sip:5201@192.168.0.205>;tag=as36bd68c6
Call-ID: 2d503e1c4a98fd3f776b447238e00600@192.168.0.205
CSeq: 101 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5201@192.168.0.205>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '280C0958-1ACC97@10.254.1.7' in 1856 ms (Method: ACK)
set_destination: Parsing <sip:2012147154@10.10.10.5:5060;isup-oli=62;2649054plex=2649054plex-elchk8eg33lcb;transport=udp> for address/port to send to
set_destination: set destination to 10.10.10.5, port 5060
Reliably Transmitting (no NAT) to 10.10.10.5:5060:
BYE sip:2012147154@10.10.10.5:5060;isup-oli=62;2649054plex=2649054plex-elchk8eg33lcb;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK520a3d5d;rport
From: <sip:5201@192.168.1.2:5060;user=phone>;tag=as40e6688f
To: "Unavailable"<sip:2012147154@10.10.10.5:5060;isup-oli=62;user=phone;2649054plex=2649054plex-elchk8eg33lcb>;tag=127.0.0.1+1+be680005+5b4d5bcb
Call-ID: 280C0958-1ACC97@10.254.1.7
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
grosvenor*CLI> sip set debug off
<--- SIP read from 10.10.10.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK520a3d5d;rport=5060
From: <sip:5201@192.168.1.2:5060;user=phone>;tag=as40e6688f
To: "Unavailable"<sip:2012147154@10.10.10.5:5060;isup-oli=62;user=phone;2649054plex=2649054plex-elchk8eg33lcb>;tag=127.0.0.1+1+be680005+5b4d5bcb
Call-ID: 280C0958-1ACC97@10.254.1.7
CSeq: 102 BYE
Server: DC-SIP/2.0
Content-Length: 0
Contact: <sip:2012147154@10.10.10.5:5060;isup-oli=62;2649054plex=2649054plex-elchk8eg33lcb;transport=udp>
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '2d503e1c4a98fd3f776b447238e00600@192.168.0.205' Method: BYE
Really destroying SIP dialog '280C0958-1ACC97@10.254.1.7' Method: ACK
grosvenor*CLI> sip set debug off