SIP Debugging Enabled for IP: 217.151.230.182:5069 call2ru*CLI> <-- SIP read from 217.151.230.182:5069: INVITE sip:111@sip.call2ru.com SIP/2.0 Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK97F00B7308C142E093AC2BB14C713B20 From: call2ru ;tag=2472302208 To: Contact: Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182 CSeq: 4072 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite build 1101 Content-Length: 145 v=0 o=120001 46618828 46618828 IN IP4 217.151.230.182 s=X-Lite c=IN IP4 217.151.230.182 t=0 0 m=audio 9000 RTP/AVP 0 a=rtpmap:0 pcmu/8000 --- (11 headers 7 lines)--- Using INVITE request as basis request - 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182 Sending to 217.151.230.182 : 5069 (NAT) Reliably Transmitting (no NAT) to 217.151.230.182:5069: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK97F00B7308C142E093AC2BB14C713B20;received=217.151.230.182 rom: call2ru ;tag=2472302208 To: ;tag=as5f63ecd1 Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182 CSeq: 4072 INVITE User-Agent: Call2Ru service Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="call2ru.com", nonce="09cd2640" Content-Length: 0 --- Scheduling destruction of call '354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182' in 15000 ms Found user '120001' call2ru*CLI> <-- SIP read from 217.151.230.182:5069: ACK sip:111@sip.call2ru.com SIP/2.0 Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK97F00B7308C142E093AC2BB14C713B20 From: call2ru ;tag=2472302208 To: ;tag=as5f63ecd1 Contact: Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182 CSeq: 4072 ACK Max-Forwards: 70 Content-Length: 0 --- (9 headers 0 lines)--- call2ru*CLI> <-- SIP read from 217.151.230.182:5069: INVITE sip:111@sip.call2ru.com SIP/2.0 Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK7C09A4CC58D542A8A22F0CD62E921DA4 From: call2ru ;tag=2472302208 To: Contact: Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182 CSeq: 4073 INVITE Proxy-Authorization: Digest username="120001",realm="call2ru.com",nonce="09cd2640",response="393ba5217012f40e958f44f 9c37e419e",uri="sip:111@sip.call2ru.com",algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite build 1101 Content-Length: 145 v=0 o=120001 46619218 46619218 IN IP4 217.151.230.182 s=X-Lite c=IN IP4 217.151.230.182 t=0 0 m=audio 9000 RTP/AVP 0 a=rtpmap:0 pcmu/8000 --- (12 headers 7 lines)--- Using INVITE request as basis request - 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182 Sending to 217.151.230.182 : 5069 (NAT) Found user '120001' Found RTP audio format 0 Peer audio RTP is at port 217.151.230.182:9000 Peer video RTP is at port 217.151.230.182:65535 Found description format pcmu Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 111 in default (domain sip.call2ru.com) list_route: hop: Transmitting (no NAT) to 217.151.230.182:5069: SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK7C09A4CC58D542A8A22F0CD62E921DA4;received=217.151.230.182 From: call2ru ;tag=2472302208 To: Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182 CSeq: 4073 INVITE User-Agent: Call2Ru service Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Answer("SIP/120001-08303fa0", "") in new stack We're at 195.224.39.103 port 15158 Video is at 195.224.39.103 port 13882 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 217.151.230.182:5069: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK7C09A4CC58D542A8A22F0CD62E921DA4;received=217.151.230.182 From: call2ru ;tag=2472302208 To: ;tag=as7602845a Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182 CSeq: 4073 INVITE User-Agent: Call2Ru service Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 162 v=0 o=root 8244 8244 IN IP4 195.224.39.103 s=session c=IN IP4 195.224.39.103 t=0 0 m=audio 15158 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- -- Executing Playback("SIP/120001-08303fa0", "tt-weasels") in new stack Sent RTP packet to 217.151.230.182:9000 (type 0, seq 19347, ts 160, len 160) -- Playing 'tt-weasels' (language 'en') call2ru*CLI> <-- SIP read from 217.151.230.182:5069: ACK sip:111@195.224.39.103 SIP/2.0 Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK79B1CC727AEB478D98AFDC157704210F From: call2ru ;tag=2472302208 To: ;tag=as7602845a Contact: Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182 CSeq: 4073 ACK Max-Forwards: 70 Content-Length: 0 --- (9 headers 0 lines)--- call2ru*CLI> <-- SIP read from 217.151.230.182:5069: BYE sip:111@195.224.39.103 SIP/2.0 Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK352A308E84E84D028454954A8AE1D4FE From: call2ru ;tag=2472302208 To: ;tag=as7602845a Contact: Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182 CSeq: 4074 BYE Max-Forwards: 70 User-Agent: X-Lite build 1101 Content-Length: 0 --- (10 headers 0 lines)--- Sending to 217.151.230.182 : 5069 (NAT) Transmitting (NAT) to 217.151.230.182:5069: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.151.230.182:5069;branch=z9hG4bK352A308E84E84D028454954A8AE1D4FE;received=217.151.230.182;rport= 5069 From: call2ru ;tag=2472302208 To: ;tag=as7602845a Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182 CSeq: 4074 BYE User-Agent: Call2Ru service Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- > cdr_odbc: Query Successful! Destroying call '354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182' call2ru*CLI> <-- SIP read from 217.151.230.182:5069: --- (0 headers 0 lines) Nat keepalive --- *** Phone receive one RTP packet (I have heared it) *** Phone send outgoing RTP, tcpdump on server saw it.