<--- SIP read from 192.168.11.123:5060 ---> INVITE sip:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK23D45EDF CSeq: 1948 INVITE To: Content-Type: application/sdp From: ;tag=857BEA0 Call-ID: 1088116812@192.168.11.123 Subject: sip:214058@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 32904 RTP/AVP 8 97 3 0 m=audio 32904 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 - 1088116812@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=z9hG4bK23D45EDF;received=192.168.11.123 From: ;tag=857BEA0 To: ;tag=as66e79e9d Call-ID: 1088116812@192.168.11.123 CSeq: 1948 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e853eec" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1088116812@192.168.11.123' in 32000 ms (Method: INVITE) Found user '214058' bee*CLI> <--- SIP read from 192.168.11.123:5060 ---> ACK sip:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK23D45EDF CSeq: 1948 ACK To: ;tag=as66e79e9d From: ;tag=857BEA0 Call-ID: 1088116812@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:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK575936FF CSeq: 1949 INVITE To: Proxy-Authorization: Digest username="214058", realm="asterisk", nonce="2e853eec", uri="sip:9214020@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="9d896546ecf93a9175e5da73dd30f45b", opaque="", algorithm="MD5" Content-Type: application/sdp From: ;tag=857BEA0 Call-ID: 1088116812@192.168.11.123 Subject: sip:214058@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 32904 RTP/AVP 8 97 3 0 m=audio 32904 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 - 1088116812@192.168.11.123 Found user '214058' 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 192.168.11.123:32904 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 - 0x8091e (gsm|ulaw|alaw|g726|g729|g726aal2|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.11.123:32904 Looking for 9214020 in local-phones (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=z9hG4bK575936FF;received=192.168.11.123 From: ;tag=857BEA0 To: Call-ID: 1088116812@192.168.11.123 CSeq: 1949 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 11280 Adding codec 0x8 (alaw) to SDP Adding codec 0x800 (g726) to SDP Adding codec 0x10 (g726aal2) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.11.120:5060: INVITE sip:214020@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK693d1714;rport From: "214058" ;tag=as6e940452 To: Contact: Call-ID: 4acf5b9012422e0332a58f136367fd90@192.168.11.124 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 10 Jul 2007 17:00:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 356 v=0 o=root 8154 8154 IN IP4 192.168.11.124 s=session c=IN IP4 192.168.11.124 t=0 0 m=audio 11280 RTP/AVP 8 111 112 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jul 10 21:00:15] WARNING[8996]: channel.c:3147 ast_channel_make_compatible: No path to translate from SIP/cisco-081edce0(256) to SIP/214058-081e8ea0(8) bee*CLI> <--- SIP read from 192.168.11.120:5060 ---> SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK693d1714;rport From: "214058" ;tag=as6e940452 To: ;tag=9FE9CD9-1F89 Date: Tue, 10 Jul 2007 17:00:13 GMT Call-ID: 4acf5b9012422e0332a58f136367fd90@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:214020@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK693d1714;rport From: "214058" ;tag=as6e940452 To: ;tag=9FE9CD9-1F89 Contact: Call-ID: 4acf5b9012422e0332a58f136367fd90@192.168.11.124 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '4acf5b9012422e0332a58f136367fd90@192.168.11.124' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 192.168.11.120:5060: BYE sip:214020@192.168.11.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK05708293;rport From: "214058" ;tag=as6e940452 To: ;tag=9FE9CD9-1F89 Call-ID: 4acf5b9012422e0332a58f136367fd90@192.168.11.124 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- 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=z9hG4bK575936FF;received=192.168.11.123 From: ;tag=857BEA0 To: ;tag=as63fc0fa6 Call-ID: 1088116812@192.168.11.123 CSeq: 1949 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Bearer capability not available X-Asterisk-HangupCauseCode: 58 <------------> bee*CLI> <--- SIP read from 192.168.11.123:5060 ---> ACK sip:9214020@192.168.11.124 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK575936FF CSeq: 1949 ACK To: ;tag=as63fc0fa6 From: ;tag=857BEA0 Call-ID: 1088116812@192.168.11.123 Content-Length: 0 User-Agent: kphone/4.2 Contact: <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '1088116812@192.168.11.123' Method: ACK bee*CLI> <--- SIP read from 192.168.11.120:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK05708293;rport From: "214058" ;tag=as6e940452 To: ;tag=9FE9CD9-1F89 Call-ID: 4acf5b9012422e0332a58f136367fd90@192.168.11.124 CSeq: 103 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [Jul 10 21:00:15] WARNING[8183]: chan_sip.c:12311 handle_response: Remote host can't match request BYE to call '4acf5b9012422e0332a58f136367fd90@192.168.11.124'. Giving up. Really destroying SIP dialog '4acf5b9012422e0332a58f136367fd90@192.168.11.124' Method: INVITE Really destroying SIP dialog '778552495@192.168.11.123' Method: REGISTER Really destroying SIP dialog '1202302177@192.168.11.123' Method: ACK