# bee*CLI> sip set debug # SIP Debugging enabled # Really destroying SIP dialog 'D1B9-1C94-46684815AD3CE939E424-003@SipHost' Method: REGISTER # bee*CLI> # <--- SIP read from 192.168.11.123:5060 ---> # INVITE sip:214019@192.168.11.124 SIP/2.0 # Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK578BC651 # CSeq: 7379 INVITE # To: # Content-Type: application/sdp # From: ;tag=44166B53 # Call-ID: 1961359474@192.168.11.123 # Subject: sip:214077@192.168.11.124 # Content-Length: 262 # User-Agent: kphone/4.2 # Contact: # # v=0 # o=username 0 0 IN IP4 192.168.11.123 # s=The Funky Flow # c=IN IP4 192.168.11.123 # t=0 0 # m=audio 32776 RTP/AVP 8 97 3 0 # m=audio 32776 RTP/AVP 97 3 0 # a=rtpmap:0 PCMU/8000 # a=rtpmap:3 GSM/8000 # a=rtpmap:8 PCMA/8000 # a=rtpmap:97 iLBC/8000 # a=fmtp:97 mode=30 # # <-------------> # --- (11 headers 12 lines) --- # Sending to 192.168.11.123 : 5060 (no NAT) # Using INVITE request as basis request - 1961359474@192.168.11.123 # # <--- Reliably Transmitting (no NAT) to 192.168.11.123:5060 ---> # SIP/2.0 407 Proxy Authentication Required # Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK578BC651;received=192.168.11.123 # From: ;tag=44166B53 # To: ;tag=as6b735b82 # Call-ID: 1961359474@192.168.11.123 # CSeq: 7379 INVITE # User-Agent: Asterisk PBX # Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY # Supported: replaces # Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="321e36d2" # Content-Length: 0 # # # <------------> # Scheduling destruction of SIP dialog '1961359474@192.168.11.123' in 32000 ms (Method: INVITE) # Found user '214077' # bee*CLI> # <--- SIP read from 192.168.11.123:5060 ---> # ACK sip:214019@192.168.11.124 SIP/2.0 # Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK578BC651 # CSeq: 7379 ACK # To: ;tag=as6b735b82 # From: ;tag=44166B53 # Call-ID: 1961359474@192.168.11.123 # Content-Length: 0 # User-Agent: kphone/4.2 # Contact: # # # <-------------> # --- (9 headers 0 lines) --- # bee*CLI> # <--- SIP read from 192.168.11.123:5060 ---> # INVITE sip:214019@192.168.11.124 SIP/2.0 # Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK46FF98B4 # CSeq: 7380 INVITE # To: # Proxy-Authorization: Digest username="214077", realm="asterisk", nonce="321e36d2", uri="sip:214019@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="72170c1418da887224977f52900e0c28", opaque="", algorithm="MD5" # Content-Type: application/sdp # From: ;tag=44166B53 # Call-ID: 1961359474@192.168.11.123 # Subject: sip:214077@192.168.11.124 # Content-Length: 262 # User-Agent: kphone/4.2 # Contact: # # v=0 # o=username 0 0 IN IP4 192.168.11.123 # s=The Funky Flow # c=IN IP4 192.168.11.123 # t=0 0 # m=audio 32776 RTP/AVP 8 97 3 0 # m=audio 32776 RTP/AVP 97 3 0 # a=rtpmap:0 PCMU/8000 # a=rtpmap:3 GSM/8000 # a=rtpmap:8 PCMA/8000 # a=rtpmap:97 iLBC/8000 # a=fmtp:97 mode=30 # # <-------------> # --- (12 headers 12 lines) --- # Sending to 192.168.11.123 : 5060 (no NAT) # Using INVITE request as basis request - 1961359474@192.168.11.123 # Found user '214077' # Found RTP audio format 8 # Found RTP audio format 97 # Found RTP audio format 3 # Found RTP audio format 0 # Found RTP audio format 97 # Found RTP audio format 3 # Found RTP audio format 0 # Peer audio RTP is at port 192.168.11.123:32776 # Found description format PCMU for ID 0 # Found description format GSM for ID 3 # Found description format PCMA for ID 8 # Found description format iLBC for ID 97 # Capabilities: us - 0x8 (alaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw) # Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) # Peer audio RTP is at port 192.168.11.123:32776 # Looking for 214019 in from-sip (domain 192.168.11.124) # list_route: hop: # # <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> # SIP/2.0 100 Trying # Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK46FF98B4;received=192.168.11.123 # rom: ;tag=44166B53 # To: # Call-ID: 1961359474@192.168.11.123 # CSeq: 7380 INVITE # User-Agent: Asterisk PBX # Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY # Supported: replaces # Contact: # Content-Length: 0 # # # <------------> # Audio is at 192.168.11.124 port 12030 # Adding codec 0x8 (alaw) to SDP # Adding codec 0x2 (gsm) to SDP # Adding codec 0x4 (ulaw) to SDP # Adding codec 0x10 (g726aal2) to SDP # Adding codec 0x20 (adpcm) to SDP # Adding codec 0x40 (slin) to SDP # Adding codec 0x80 (lpc10) to SDP # Adding codec 0x100 (g729) to SDP # Adding codec 0x400 (ilbc) to SDP # Adding codec 0x800 (g726) to SDP # Adding non-codec 0x1 (telephone-event) to SDP # Reliably Transmitting (no NAT) to 192.168.11.120:5060: # INVITE sip:214019@192.168.11.120 SIP/2.0 # Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240a0997;rport # From: "214077" ;tag=as5289d0d2 # To: # Contact: # Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124 # CSeq: 102 INVITE # User-Agent: Asterisk PBX # Max-Forwards: 70 # Date: Tue, 09 Oct 2007 10:41:39 GMT # Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY # Supported: replaces # Content-Type: application/sdp # Content-Length: 522 # # v=0 # o=root 21640 21640 IN IP4 192.168.11.124 # s=session # c=IN IP4 192.168.11.124 # t=0 0 # m=audio 12030 RTP/AVP 8 3 0 112 5 10 7 18 97 111 101 # a=rtpmap:8 PCMA/8000 # a=rtpmap:3 GSM/8000 # a=rtpmap:0 PCMU/8000 # a=rtpmap:112 AAL2-G726-32/8000 # a=rtpmap:5 DVI4/8000 # a=rtpmap:10 L16/8000 # a=rtpmap:7 LPC/8000 # a=rtpmap:18 G729/8000 # a=fmtp:18 annexb=no # a=rtpmap:97 iLBC/8000 # a=fmtp:97 mode=30 # a=rtpmap:111 G726-32/8000 # a=rtpmap:101 telephone-event/8000 # a=fmtp:101 0-16 # a=silenceSupp:off - - - - # a=ptime:20 # a=sendrecv # # --- # bee*CLI> # <--- SIP read from 192.168.11.120:5060 ---> # SIP/2.0 100 Trying # Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240a0997;rport # From: "214077" ;tag=as5289d0d2 # To: ;tag=DE34DCED-2550 # Date: Tue, 09 Oct 2007 10:40:50 GMT # Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124 # Server: Cisco-SIPGateway/IOS-12.x # CSeq: 102 INVITE # Allow-Events: telephone-event # Content-Length: 0 # # # <-------------> # --- (10 headers 0 lines) --- # bee*CLI> # <--- SIP read from 192.168.11.120:5060 ---> # SIP/2.0 183 Session Progress # Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240a0997;rport # From: "214077" ;tag=as5289d0d2 # To: ;tag=DE34DCED-2550 # Date: Tue, 09 Oct 2007 10:40:50 GMT # Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124 # Server: Cisco-SIPGateway/IOS-12.x # CSeq: 102 INVITE # Allow-Events: telephone-event # Contact: # Content-Disposition: session;handling=required # Content-Type: application/sdp # Content-Length: 252 # # v=0 # o=CiscoSystemsSIP-GW-UserAgent 8301 156 IN IP4 192.168.11.120 # s=SIP Call # c=IN IP4 192.168.11.120 # t=0 0 # m=audio 18898 RTP/AVP 8 101 # c=IN IP4 192.168.11.120 # a=rtpmap:8 PCMA/8000 # a=rtpmap:101 telephone-event/8000 # a=fmtp:101 0-16 # a=ptime:20 # # <-------------> # --- (13 headers 11 lines) --- # Found RTP audio format 8 # Found RTP audio format 101 # Peer audio RTP is at port 192.168.11.120:18898 # Found description format PCMA for ID 8 # Found description format telephone-event for ID 101 # Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) # Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) # Peer audio RTP is at port 192.168.11.120:18898 # Audio is at 192.168.11.124 port 16230 # Adding codec 0x8 (alaw) to SDP # # <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> # SIP/2.0 183 Session Progress # Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK46FF98B4;received=192.168.11.123 # From: ;tag=44166B53 # To: ;tag=as0353e318 # Call-ID: 1961359474@192.168.11.123 # CSeq: 7380 INVITE # User-Agent: Asterisk PBX # Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY # Supported: replaces # Contact: # Content-Type: application/sdp # Content-Length: 188 # # v=0 # o=root 21640 21640 IN IP4 192.168.11.124 # s=session # c=IN IP4 192.168.11.124 # t=0 0 # m=audio 16230 RTP/AVP 8 # a=rtpmap:8 PCMA/8000 # a=silenceSupp:off - - - - # a=ptime:20 # a=sendrecv # # <------------> # Really destroying SIP dialog 'D1B9-1C94-4668481595DDF37A5E3A-004@SipHost' Method: REGISTER # bee*CLI> # <--- SIP read from 192.168.11.128:58559 ---> # INVITE sip:1001@192.168.11.124:5060 SIP/2.0 # Via: SIP/2.0/UDP 192.168.11.128:5060 # From: "unknown" ;tag=197FC956-133B # To: # Date: Fri, 05 Mar 1993 22:50:04 GMT # Call-ID: 976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128 # Supported: timer,100rel # Min-SE: 1800 # Cisco-Guid: 2539994139-416420300-2150403679-25483487 # User-Agent: Cisco-SIPGateway/IOS-12.x # Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO # CSeq: 101 INVITE # Max-Forwards: 6 # Remote-Party-ID: "unknown" ;party=calling;screen=no;privacy=off # Timestamp: 731371804 # Contact: # Expires: 180 # Allow-Events: telephone-event # Content-Type: application/sdp # Content-Length: 287 # # v=0 # o=CiscoSystemsSIP-GW-UserAgent 3633 652 IN IP4 192.168.11.128 # s=SIP Call # c=IN IP4 192.168.11.128 # t=0 0 # m=audio 19016 RTP/AVP 18 8 101 # c=IN IP4 192.168.11.128 # a=rtpmap:18 G729/8000 # a=fmtp:18 annexb=no # a=rtpmap:8 PCMA/8000 # a=rtpmap:101 telephone-event/8000 # a=fmtp:101 0-16 # # <-------------> # --- (20 headers 12 lines) --- # Sending to 192.168.11.128 : 5060 (no NAT) # Using INVITE request as basis request - 976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128 # Found no matching peer or user for '192.168.11.128:58559' # [Oct 9 14:41:41] NOTICE[21665]: chan_sip.c:13587 handle_request_invite: Failed to authenticate user "unknown" ;tag=197FC956-133B # # <--- Reliably Transmitting (no NAT) to 192.168.11.128:5060 ---> # SIP/2.0 403 Forbidden # Via: SIP/2.0/UDP 192.168.11.128:5060;received=192.168.11.128 # From: "unknown" ;tag=197FC956-133B # To: ;tag=as1aed72aa # Call-ID: 976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128 # CSeq: 101 INVITE # User-Agent: Asterisk PBX # Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY # Supported: replaces # Content-Length: 0 # # # <------------> # Scheduling destruction of SIP dialog '976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128' in 32000 ms (Method: INVITE) # bee*CLI> # <--- SIP read from 192.168.11.128:58559 ---> # ACK sip:1001@192.168.11.124:5060 SIP/2.0 # Via: SIP/2.0/UDP 192.168.11.128:5060 # From: "unknown" ;tag=197FC956-133B # To: ;tag=as1aed72aa # Date: Fri, 05 Mar 1993 22:50:04 GMT # Call-ID: 976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128 # Max-Forwards: 6 # Content-Length: 0 # CSeq: 101 ACK # # # <-------------> # --- (9 headers 0 lines) --- # Really destroying SIP dialog '976F9BF3-18D211CC-80308E5F-184D8DF@192.168.11.128' Method: ACK # bee*CLI> # <--- SIP read from 192.168.11.123:5060 ---> # CANCEL sip:214019@192.168.11.124 SIP/2.0 # Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK46FF98B4 # CSeq: 7380 CANCEL # To: # From: ;tag=44166B53 # Call-ID: 1961359474@192.168.11.123 # Content-Length: 0 # User-Agent: kphone/4.2 # Contact: # # # <-------------> # --- (9 headers 0 lines) --- # Sending to 192.168.11.123 : 5060 (no NAT) # # <--- Reliably Transmitting (no NAT) to 192.168.11.123:5060 ---> # SIP/2.0 487 Request Terminated # Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK46FF98B4;received=192.168.11.123 # From: ;tag=44166B53 # To: ;tag=as0353e318 # Call-ID: 1961359474@192.168.11.123 # CSeq: 7380 INVITE # User-Agent: Asterisk PBX # Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY # Supported: replaces # Content-Length: 0 # # # <------------> # # <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> # SIP/2.0 200 OK # Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK46FF98B4;received=192.168.11.123 # From: ;tag=44166B53 # To: ;tag=as0353e318 # Call-ID: 1961359474@192.168.11.123 # CSeq: 7380 CANCEL # User-Agent: Asterisk PBX # Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY # Supported: replaces # Contact: # Content-Length: 0 # # # <------------> # Scheduling destruction of SIP dialog '2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124' in 32000 ms (Method: INVITE) # Reliably Transmitting (no NAT) to 192.168.11.120:5060: # CANCEL sip:214019@192.168.11.120 SIP/2.0 # Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240a0997;rport # From: "214077" ;tag=as5289d0d2 # To: # Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124 # CSeq: 102 CANCEL # User-Agent: Asterisk PBX # Max-Forwards: 70 # Content-Length: 0 # # # --- # Scheduling destruction of SIP dialog '2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124' in 32000 ms (Method: INVITE) # bee*CLI> # <--- SIP read from 192.168.11.120:5060 ---> # SIP/2.0 200 OK # Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240a0997;rport # From: "214077" ;tag=as5289d0d2 # To: # Date: Tue, 09 Oct 2007 10:40:54 GMT # Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124 # Content-Length: 0 # CSeq: 102 CANCEL # # # <-------------> # --- (8 headers 0 lines) --- # bee*CLI> # <--- SIP read from 192.168.11.120:5060 ---> # SIP/2.0 487 Request Cancelled # Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240a0997;rport # From: "214077" ;tag=as5289d0d2 # To: ;tag=DE34DCED-2550 # Date: Tue, 09 Oct 2007 10:40:54 GMT # Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124 # Server: Cisco-SIPGateway/IOS-12.x # CSeq: 102 INVITE # Allow-Events: telephone-event # Content-Length: 0 # # # <-------------> # --- (10 headers 0 lines) --- # Transmitting (no NAT) to 192.168.11.120:5060: # ACK sip:214019@192.168.11.120 SIP/2.0 # Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240a0997;rport # From: "214077" ;tag=as5289d0d2 # To: ;tag=DE34DCED-2550 # Contact: # Call-ID: 2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124 # CSeq: 102 ACK # User-Agent: Asterisk PBX # Max-Forwards: 70 # Content-Length: 0 # # # --- # Really destroying SIP dialog '2efa2dfe12e18f220b3f11c47102be7f@192.168.11.124' Method: INVITE # bee*CLI> # <--- SIP read from 192.168.11.123:5060 ---> # ACK sip:214019@192.168.11.124 SIP/2.0 # Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK46FF98B4 # CSeq: 7380 ACK # From: ;tag=44166B53 # To: ;tag=as0353e318 # Call-ID: 1961359474@192.168.11.123 # Content-Length: 0 # User-Agent: kphone/4.2 # Contact: # # # <-------------> # --- (9 headers 0 lines) --- # Really destroying SIP dialog '1961359474@192.168.11.123' Method: ACK # bee*CLI> sip no debug # SIP Debugging Disabled # bee*CLI>