DEBUG[114703]: File manager.c, Line 573 (process_message): Manager received command 'originate' DEBUG[114703]: File chan_sip.c, Line 656 (create_addr): Setting NAT on RTP to 0 DEBUG[114703]: File chan_sip.c, Line 856 (sip_call): Outgoing Call for Jeremy DEBUG[114703]: File chan_sip.c, Line 943 (find_user): Jeremy is not a local user We're at 192.168.1.1 port 10104 Answering with preferred capability 4 Answering with non-codec capability 1 11 headers, 9 lines Reliably Transmitting: INVITE sip:Jeremy@192.168.1.169 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK4c0e913a From: "asterisk" ;tag=as30dfa3cb To: Contact: Call-ID: 58db94e9174804345b151e12691a7219@192.168.1.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 187 v=0 o=root 11621 11621 IN IP4 192.168.1.1 s=session c=IN IP4 192.168.1.1 t=0 0 m=audio 10104 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 192.168.1.169:5060 DEBUG[114703]: File chan_sip.c, Line 1013 (sip_hangup): find_user(Jeremy) - decrement outUse counter DEBUG[114703]: File chan_sip.c, Line 943 (find_user): Jeremy is not a local user Reliably Transmitting: CANCEL sip:Jeremy@192.168.1.169 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK4c0e913a From: "asterisk" ;tag=as30dfa3cb To: Contact: Call-ID: 58db94e9174804345b151e12691a7219@192.168.1.1 CSeq: 102 CANCEL User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.1.169:5060 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK4c0e913a From: "asterisk" ;tag=as30dfa3cb To: ;tag=28210800bf02c7c50ed2827-610d4a66 Call-ID: 58db94e9174804345b151e12691a7219@192.168.1.1 Server: Cisco-SIP-IP-Phone/2 CSeq: 102 INVITE Content-Length: 0 8 headers, 0 lines DEBUG[40966]: File chan_sip.c, Line 567 (__sip_semi_ack): (Provisional) Stopping retransmission (but retaining packet) on '58db94e9174804345b151e12691a7219@192.168.1.1' Request 102: Found Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK4c0e913a From: "asterisk" ;tag=as30dfa3cb To: ;tag=28210800bf02c7c50ed2827-610d4a66 Call-ID: 58db94e9174804345b151e12691a7219@192.168.1.1 Server: Cisco-SIP-IP-Phone/2 CSeq: 102 INVITE Content-Length: 0 8 headers, 0 lines DEBUG[40966]: File chan_sip.c, Line 567 (__sip_semi_ack): (Provisional) Stopping retransmission (but retaining packet) on '58db94e9174804345b151e12691a7219@192.168.1.1' Request 102: Found Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK4c0e913a From: "asterisk" ;tag=as30dfa3cb To: ;tag=28210800bf02c7c50ed2827-610d4a66 Call-ID: 58db94e9174804345b151e12691a7219@192.168.1.1 Server: Cisco-SIP-IP-Phone/2 CSeq: 102 CANCEL Content-Length: 0 8 headers, 0 lines DEBUG[40966]: File chan_sip.c, Line 529 (__sip_ack): Acked pending invite 102 DEBUG[40966]: File chan_sip.c, Line 547 (__sip_ack): Stopping retransmission on '58db94e9174804345b151e12691a7219@192.168.1.1' of Request 102: Found Sip read: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK4c0e913a From: "asterisk" ;tag=as30dfa3cb To: ;tag=28210800bf02c7c50ed2827-610d4a66 Call-ID: 58db94e9174804345b151e12691a7219@192.168.1.1 Server: Cisco-SIP-IP-Phone/2 CSeq: 102 INVITE Content-Length: 0 8 headers, 0 lines DEBUG[40966]: File chan_sip.c, Line 547 (__sip_ack): Stopping retransmission on '58db94e9174804345b151e12691a7219@192.168.1.1' of Request 102: Found Transmitting: ACK sip:Jeremy@192.168.1.169 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK4c0e913a From: "asterisk" ;tag=as30dfa3cb To: ;tag=28210800bf02c7c50ed2827-610d4a66 Contact: Call-ID: 58db94e9174804345b151e12691a7219@192.168.1.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.1.169:5060 DEBUG[40966]: File chan_sip.c, Line 872 (__sip_destroy): Destorying call '58db94e9174804345b151e12691a7219@192.168.1.1'