ee*CLI> <--- SIP read from 192.168.11.123:5062 ---> INVITE sip:693977@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK9A26C9F CSeq: 390 INVITE To: Content-Type: application/sdp From: ;tag=2D4F9484 Call-ID: 1665297040@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 32782 RTP/AVP 8 97 3 0 m=audio 32782 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 : 5062 (no NAT) Using INVITE request as basis request - 1665297040@192.168.11.123 <--- Reliably Transmitting (no NAT) to 192.168.11.123:5062 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK9A26C9F;received=192.168.11.123 From: ;tag=2D4F9484 To: ;tag=as77643dba Call-ID: 1665297040@192.168.11.123 CSeq: 390 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d44f613" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1665297040@192.168.11.123' in 32000 ms (Method: INVITE) Found user '214077' bee*CLI> <--- SIP read from 192.168.11.123:5062 ---> ACK sip:693977@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK9A26C9F CSeq: 390 ACK To: ;tag=as77643dba From: ;tag=2D4F9484 Call-ID: 1665297040@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Contact: <-------------> --- (9 headers 0 lines) --- bee*CLI> <--- SIP read from 192.168.11.123:5062 ---> INVITE sip:693977@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK1C2782E9 CSeq: 391 INVITE To: Proxy-Authorization: Digest username="214077", realm="asterisk", nonce="3d44f613", uri="sip:693977@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="4c2041fc1e939228ce2a67958c9ff4a7", opaque="", algorithm="MD5" Content-Type: application/sdp From: ;tag=2D4F9484 Call-ID: 1665297040@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 32782 RTP/AVP 8 97 3 0 m=audio 32782 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 : 5062 (no NAT) Using INVITE request as basis request - 1665297040@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 192.168.11.123:32782 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 192.168.11.123:32782 Looking for 693977 in from-sip (domain 192.168.11.124) list_route: hop: <--- Transmitting (no NAT) to 192.168.11.123:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK1C2782E9;received=192.168.11.123 From: ;tag=2D4F9484 To: Call-ID: 1665297040@192.168.11.123 CSeq: 391 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 15138 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.11.128:5060: INVITE sip:693977@192.168.11.128 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK1549241c;rport From: "214077" ;tag=as39d21cfe To: Contact: Call-ID: 0f35ed3461aa30c61e108d723fb794d6@192.168.11.124 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 11 Oct 2007 13:23:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 244 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 15138 RTP/AVP 8 101 a=rtpmap:8 PCMA/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 192.168.11.128:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK1549241c;rport From: "214077" ;tag=as39d21cfe To: ;tag=246DD5-1A40 Date: Mon, 01 Mar 1993 00:39:47 GMT Call-ID: 0f35ed3461aa30c61e108d723fb794d6@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 192.168.11.128:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK1549241c;rport From: "214077" ;tag=as39d21cfe To: ;tag=246DD5-1A40 Date: Mon, 01 Mar 1993 00:39:47 GMT Call-ID: 0f35ed3461aa30c61e108d723fb794d6@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 192.168.11.128:5060: ACK sip:693977@192.168.11.128 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK1549241c;rport From: "214077" ;tag=as39d21cfe To: ;tag=246DD5-1A40 Contact: Call-ID: 0f35ed3461aa30c61e108d723fb794d6@192.168.11.124 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- bee*CLI> <--- Transmitting (no NAT) to 192.168.11.123:5062 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK1C2782E9;received=192.168.11.123 From: ;tag=2D4F9484 To: ;tag=as0592e4e8 Call-ID: 1665297040@192.168.11.123 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Circuit/channel congestion X-Asterisk-HangupCauseCode: 34 <------------> bee*CLI> <--- SIP read from 192.168.11.123:5062 ---> ACK sip:693977@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK1C2782E9 CSeq: 391 ACK To: ;tag=as0592e4e8 From: ;tag=2D4F9484 Call-ID: 1665297040@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Contact: <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '0f35ed3461aa30c61e108d723fb794d6@192.168.11.124' Method: INVITE Really destroying SIP dialog '1665297040@192.168.11.123' Method: ACK bee*CLI>