# 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";tag=127.0.0.1+1+be680005+5b4d5bcb To: 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" Privacy: none Max-Forwards: 56 Contact: "Unavailable" 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: <--- 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";tag=127.0.0.1+1+be680005+5b4d5bcb To: 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: 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" ;tag=as36bd68c6 To: Contact: 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" ;tag=as36bd68c6 To: Call-ID: 2d503e1c4a98fd3f776b447238e00600@192.168.0.205 Date: Wed, 13 Jun 2007 GMT CSeq: 102 INVITE Server: Cisco-CP7961G/8.0 Contact: 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" ;tag=as36bd68c6 To: ;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: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "Eric Lubow" ;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";tag=127.0.0.1+1+be680005+5b4d5bcb To: ;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: 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" ;tag=as36bd68c6 To: ;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: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "Eric Lubow" ;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: set_destination: Parsing 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" ;tag=as36bd68c6 To: ;tag=0019e8f4b4d0181a883d0369-e4a7fe5e Contact: 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";tag=127.0.0.1+1+be680005+5b4d5bcb To: ;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: 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";tag=127.0.0.1+1+be680005+5b4d5bcb To: ;tag=as40e6688f CSeq: 761307750 ACK Contact: "Unavailable" 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: ;tag=0019e8f4b4d0181a883d0369-e4a7fe5e To: "Unavailable" ;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: ;tag=0019e8f4b4d0181a883d0369-e4a7fe5e To: "Unavailable" ;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: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '280C0958-1ACC97@10.254.1.7' in 1856 ms (Method: ACK) set_destination: Parsing 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: ;tag=as40e6688f To: "Unavailable";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: ;tag=as40e6688f To: "Unavailable";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: <-------------> --- (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