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=z9hG4bK4DFAA20E CSeq: 2134 INVITE To: Content-Type: application/sdp From: ;tag=23DB806F Call-ID: 547201740@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 32774 RTP/AVP 8 97 3 0 m=audio 32774 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 - 547201740@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=z9hG4bK4DFAA20E;received=192.168.11.123 From: ;tag=23DB806F To: ;tag=as37f1c9b2 Call-ID: 547201740@192.168.11.123 CSeq: 2134 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="606c3fbb" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '547201740@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=z9hG4bK4DFAA20E CSeq: 2134 ACK To: ;tag=as37f1c9b2 From: ;tag=23DB806F Call-ID: 547201740@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=z9hG4bK2A48836A CSeq: 2135 INVITE To: Proxy-Authorization: Digest username="214077", realm="asterisk", nonce="606c3fbb", uri="sip:214019@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="e5a8ab65ca6f335eb27af7ed617923c1", opaque="", algorithm="MD5" Content-Type: application/sdp From: ;tag=23DB806F Call-ID: 547201740@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 32774 RTP/AVP 8 97 3 0 m=audio 32774 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 - 547201740@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 192.168.11.123:32774 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 192.168.11.123:32774 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=z9hG4bK2A48836A;received=192.168.11.123 From: ;tag=23DB806F To: Call-ID: 547201740@192.168.11.123 CSeq: 2135 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 15878 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=z9hG4bK240f8219;rport From: "214077" ;tag=as56c918c2 To: Contact: Call-ID: 4626a4955f6eb78915d45c1922f57057@192.168.11.124 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 09 Oct 2007 10:41:21 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 15878 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=z9hG4bK240f8219;rport From: "214077" ;tag=as56c918c2 To: ;tag=DE34962C-14F4 Date: Tue, 09 Oct 2007 10:40:32 GMT Call-ID: 4626a4955f6eb78915d45c1922f57057@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/2.0 486 Busy here Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK240f8219;rport From: "214077" ;tag=as56c918c2 To: ;tag=DE34962C-14F4 Date: Tue, 09 Oct 2007 10:40:32 GMT Call-ID: 4626a4955f6eb78915d45c1922f57057@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=z9hG4bK240f8219;rport From: "214077" ;tag=as56c918c2 To: ;tag=DE34962C-14F4 Contact: Call-ID: 4626a4955f6eb78915d45c1922f57057@192.168.11.124 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- bee*CLI> <--- Transmitting (no NAT) to 192.168.11.123:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK2A48836A;received=192.168.11.123 From: ;tag=23DB806F To: ;tag=as1af642f3 Call-ID: 547201740@192.168.11.123 CSeq: 2135 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: User busy X-Asterisk-HangupCauseCode: 17 <------------> 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=z9hG4bK2A48836A CSeq: 2135 ACK To: ;tag=as1af642f3 From: ;tag=23DB806F Call-ID: 547201740@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Contact: <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '4626a4955f6eb78915d45c1922f57057@192.168.11.124' Method: INVITE Really destroying SIP dialog '547201740@192.168.11.123' Method: ACK Really destroying SIP dialog '1877302052@192.168.11.123' Method: REGISTER bee*CLI> <--- SIP read from 192.168.11.120:56021 ---> BYE sip:216@192.168.11.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.120:5060 From: ;tag=DE32DA63-10E6 To: "216" ;tag=as1d1d919c Date: Tue, 09 Oct 2007 10:38:45 GMT Call-ID: 29d964f528d6bfb678c6964a2e37bd47@192.168.11.124 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 6 Timestamp: 1191926437 CSeq: 101 BYE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.11.120 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.11.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.11.120:5060;received=192.168.11.120 From: ;tag=DE32DA63-10E6 To: "216" ;tag=as1d1d919c Call-ID: 29d964f528d6bfb678c6964a2e37bd47@192.168.11.124 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 'D1B9-1C94-47111489E1C3E56C1892-222@SipHost' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.11.131, port 5060 Reliably Transmitting (no NAT) to 192.168.11.131:5060: BYE sip:216@192.168.11.131:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK0809cd3a;rport From: ;tag=as2381c048 To: "216" ;tag=8f34d32b-111489 Call-ID: D1B9-1C94-47111489E1C3E56C1892-222@SipHost CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- bee*CLI> <--- SIP read from 192.168.11.131:5060 ---> SIP/2.0 200 OK Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK0809cd3a From:;tag=as2381c048 To:"216" ;tag=8f34d32b-111489 Call-ID:D1B9-1C94-47111489E1C3E56C1892-222@SipHost CSeq:102 BYE User-Agent:dlink 12-36-7924821 Content-Length:0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '29d964f528d6bfb678c6964a2e37bd47@192.168.11.124' Method: BYE Really destroying SIP dialog 'D1B9-1C94-47111489E1C3E56C1892-222@SipHost' Method: ACK bee*CLI> <--- SIP read from 192.168.11.131:5060 ---> BYE sip:9511804@192.168.11.124 SIP/2.0 Via:SIP/2.0/UDP 192.168.11.131:5060;branch=z9hG4bK23964c042b2642e6 From:"216" ;tag=8f34d32b-111489 To:;tag=as2381c048 Call-ID:D1B9-1C94-47111489E1C3E56C1892-222@SipHost CSeq:6018 BYE Max-Forwards:70 User-Agent:dlink 12-36-7924821 Content-Length:0 <-------------> --- (9 headers 0 lines) --- <--- Transmitting (no NAT) to 192.168.11.131:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.11.131:5060;branch=z9hG4bK23964c042b2642e6;received=192.168.11.131 From: "216" ;tag=8f34d32b-111489 To: ;tag=as2381c048 Call-ID: D1B9-1C94-47111489E1C3E56C1892-222@SipHost CSeq: 6018 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Really destroying SIP dialog 'D1B9-1C94-466848152998B392D344-001@SipHost' Method: REGISTER Really destroying SIP dialog 'D1B9-1C94-466848153EC9999E4BE8-002@SipHost' Method: REGISTER bee*CLI> sip no debug SIP Debugging Disabled 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>