<------------> Scheduling destruction of SIP dialog '7ae049f51b17b2d930ded27474cad6bc@192.168.11.129' in 32000 ms (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=z9hG4bK4469F3DD CSeq: 3872 INVITE To: Content-Type: application/sdp From: ;tag=225F6D3F Call-ID: 250702825@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 32804 RTP/AVP 8 97 3 0 m=audio 32804 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 - 250702825@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=z9hG4bK4469F3DD;received=192.168.11.123 From: ;tag=225F6D3F To: ;tag=as233d372d Call-ID: 250702825@192.168.11.123 CSeq: 3872 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a2952bc" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '250702825@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=z9hG4bK4469F3DD CSeq: 3872 ACK To: ;tag=as233d372d From: ;tag=225F6D3F Call-ID: 250702825@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=z9hG4bK5CE8F01F CSeq: 3873 INVITE To: Proxy-Authorization: Digest username="214077", realm="asterisk", nonce="3a2952bc", uri="sip:214019@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="f3424eddcd6b8e09480b32fd8225b8d6", opaque="", algorithm="MD5" Content-Type: application/sdp From: ;tag=225F6D3F Call-ID: 250702825@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 32804 RTP/AVP 8 97 3 0 m=audio 32804 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 - 250702825@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 192.168.11.123:32804 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 192.168.11.123:32804 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=z9hG4bK5CE8F01F;received=192.168.11.123 From: ;tag=225F6D3F To: Call-ID: 250702825@192.168.11.123 CSeq: 3873 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 13684 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=z9hG4bK5f1284ea;rport From: "214077" ;tag=as03a357bf To: Contact: Call-ID: 392f8c2d588756633723704446b7fbd2@192.168.11.124 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Oct 2007 12:54:33 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 13684 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=z9hG4bK5f1284ea;rport From: "214077" ;tag=as03a357bf To: ;tag=B57BCEB1-440 Date: Mon, 01 Oct 2007 12:53:48 GMT Call-ID: 392f8c2d588756633723704446b7fbd2@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=z9hG4bK5f1284ea;rport From: "214077" ;tag=as03a357bf To: ;tag=B57BCEB1-440 Date: Mon, 01 Oct 2007 12:53:48 GMT Call-ID: 392f8c2d588756633723704446b7fbd2@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: 253 v=0 o=CiscoSystemsSIP-GW-UserAgent 1454 1359 IN IP4 192.168.11.120 s=SIP Call c=IN IP4 192.168.11.120 t=0 0 m=audio 18412 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 192.168.11.120:18412 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 192.168.11.120:18412 Audio is at 192.168.11.124 port 16638 Adding codec 0x8 (alaw) to SDP bee*CLI> <--- 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=z9hG4bK5CE8F01F;received=192.168.11.123 From: ;tag=225F6D3F To: ;tag=as4b563b6c Call-ID: 250702825@192.168.11.123 CSeq: 3873 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 16638 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> 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=z9hG4bK5CE8F01F CSeq: 3873 CANCEL To: From: ;tag=225F6D3F Call-ID: 250702825@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=z9hG4bK5CE8F01F;received=192.168.11.123 From: ;tag=225F6D3F To: ;tag=as4b563b6c Call-ID: 250702825@192.168.11.123 CSeq: 3873 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=z9hG4bK5CE8F01F;received=192.168.11.123 From: ;tag=225F6D3F To: ;tag=as4b563b6c Call-ID: 250702825@192.168.11.123 CSeq: 3873 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 '392f8c2d588756633723704446b7fbd2@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=z9hG4bK5f1284ea;rport From: "214077" ;tag=as03a357bf To: Call-ID: 392f8c2d588756633723704446b7fbd2@192.168.11.124 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '392f8c2d588756633723704446b7fbd2@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=z9hG4bK5f1284ea;rport From: "214077" ;tag=as03a357bf To: Date: Mon, 01 Oct 2007 12:53:50 GMT Call-ID: 392f8c2d588756633723704446b7fbd2@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=z9hG4bK5f1284ea;rport From: "214077" ;tag=as03a357bf To: ;tag=B57BCEB1-440 Date: Mon, 01 Oct 2007 12:53:50 GMT Call-ID: 392f8c2d588756633723704446b7fbd2@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=z9hG4bK5f1284ea;rport From: "214077" ;tag=as03a357bf To: ;tag=B57BCEB1-440 Contact: Call-ID: 392f8c2d588756633723704446b7fbd2@192.168.11.124 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '392f8c2d588756633723704446b7fbd2@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=z9hG4bK5CE8F01F CSeq: 3873 ACK From: ;tag=225F6D3F To: ;tag=as4b563b6c Call-ID: 250702825@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Contact: <------------->