*CLI> <--- SIP read from UDP:192.168.10.33:5065 ---> INVITE sip:123@192.168.10.43 SIP/2.0 Date: Mon, 06 Jun 2011 07:01:27 GMT CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bK32f97d77-788e-e011-9ec0-d48564eb06ac;rport User-Agent: Ekiga/2.0.12 From: "max lit" ;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer To: Contact: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Type: application/sdp Content-Length: 206 Max-Forwards: 70 v=0 o=- 1307343687 1307343687 IN IP4 192.168.10.33 s=Opal SIP Session c=IN IP4 192.168.10.33 t=0 0 m=audio 5004 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 9 lines) --- Sending to 192.168.10.33:5065 (no NAT) Using INVITE request as basis request - 24084b77-788e-e011-9ec0-d48564eb06ac@explorer Found peer '192.168.10.33' for 'litnimax' from 192.168.10.33:5065 <--- Reliably Transmitting (no NAT) to 192.168.10.33:5065 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bK32f97d77-788e-e011-9ec0-d48564eb06ac;received=192.168.10.33;rport=5065 From: "max lit" ;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac To: ;tag=as1ebd5092 Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer CSeq: 1 INVITE Server: Asterisk PBX 1.8.4.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5db2c25b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '24084b77-788e-e011-9ec0-d48564eb06ac@explorer' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.10.33:5065 ---> ACK sip:123@192.168.10.43 SIP/2.0 CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bK32f97d77-788e-e011-9ec0-d48564eb06ac;rport From: "max lit" ;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer To: ;tag=as1ebd5092 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Length: 0 Max-Forwards: 70 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.10.33:5065 ---> INVITE sip:123@192.168.10.43 SIP/2.0 Date: Mon, 06 Jun 2011 07:01:27 GMT CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bKca327f77-788e-e011-9ec0-d48564eb06ac;rport User-Agent: Ekiga/2.0.12 Authorization: Digest username="192.168.10.33", realm="asterisk", nonce="5db2c25b", uri="sip:123@192.168.10.43", algorithm=md5, response="c97071284031081c32b827927bcf0482" From: "max lit" ;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer To: Contact: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Type: application/sdp Content-Length: 206 Max-Forwards: 70 v=0 o=- 1307343687 1307343687 IN IP4 192.168.10.33 s=Opal SIP Session c=IN IP4 192.168.10.33 t=0 0 m=audio 5004 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (14 headers 9 lines) --- Sending to 192.168.10.33:5065 (no NAT) Using INVITE request as basis request - 24084b77-788e-e011-9ec0-d48564eb06ac@explorer Found peer '192.168.10.33' for '192.168.10.33' from 192.168.10.33:5065 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 192.168.10.33:5004 Looking for 123 in from-max (domain 192.168.10.43) <--- Reliably Transmitting (no NAT) to 192.168.10.33:5065 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bKca327f77-788e-e011-9ec0-d48564eb06ac;received=192.168.10.33;rport=5065 From: "max lit" ;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac To: ;tag=as1ebd5092 Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer CSeq: 2 INVITE Server: Asterisk PBX 1.8.4.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jun 6 11:00:53] NOTICE[22766]: chan_sip.c:21619 handle_request_invite: Call from 'maxauth' to extension '123' rejected because extension not found in context 'from-max'. Scheduling destruction of SIP dialog '24084b77-788e-e011-9ec0-d48564eb06ac@explorer' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.10.33:5065 ---> ACK sip:123@192.168.10.43 SIP/2.0 CSeq: 2 ACK Via: SIP/2.0/UDP 192.168.10.33:5065;branch=z9hG4bKca327f77-788e-e011-9ec0-d48564eb06ac;rport Authorization: Digest username="192.168.10.33", realm="asterisk", nonce="5db2c25b", uri="sip:123@192.168.10.43", algorithm=md5, response="0d24bb0b38c42908bc292abd66a94cb8" From: "max lit" ;tag=f40a4b77-788e-e011-9ec0-d48564eb06ac Call-ID: 24084b77-788e-e011-9ec0-d48564eb06ac@explorer To: ;tag=as1ebd5092 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Length: 0 Max-Forwards: 70 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog 'de290942-788e-e011-9ec0-d48564eb06ac@explorer' Method: REGISTER Really destroying SIP dialog '24084b77-788e-e011-9ec0-d48564eb06ac@explorer' Method: ACK