? ? ???????? ?????? ?? ???????? ??????, ??? ???? ??? ???????? ? ???????? <-- SIP read from 192.168.1.43:5060: INVITE sip:74957757566@192.168.1.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.43;rport;branch=z9hG4bKncdolnst Max-Forwards: 70 To: From: "201" ;tag=isiyh Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 962 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.1 Content-Length: 304 v=0 o=201 1770785376 570897782 IN IP4 192.168.1.43 s=- c=IN IP4 192.168.1.43 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 --- (13 headers 14 lines) --- Using INVITE request as basis request - cjlqmdbgmjanzth@192.168.1.43 Sending to 192.168.1.43 : 5060 (NAT) Reliably Transmitting (no NAT) to 192.168.1.43:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.43;branch=z9hG4bKncdolnst;received=192.168.1.43;rport=5060 From: "201" ;tag=isiyh To: ;tag=as197c8170 Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 962 INVITE User-Agent: Usr Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38433a2b" Content-Length: 0 --- Scheduling destruction of call 'cjlqmdbgmjanzth@192.168.1.43' in 15000 ms Found user '201' <-- SIP read from 192.168.1.43:5060: ACK sip:74957757566@192.168.1.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.43;rport;branch=z9hG4bKncdolnst Max-Forwards: 70 To: ;tag=as197c8170 From: "201" ;tag=isiyh Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 962 ACK User-Agent: Twinkle/1.1 Content-Length: 0 --- (9 headers 0 lines) --- <-- SIP read from 192.168.1.43:5060: INVITE sip:74957757566@192.168.1.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.43;rport;branch=z9hG4bKsyufaoni Max-Forwards: 70 Proxy-Authorization: Digest username="201",realm="asterisk",nonce="38433a2b",uri="sip:74957757566@192.168.1.6",response="a5f436e27fd3e6e3fb7044eb0fc485c6",algorithm=MD5 To: From: "201" ;tag=isiyh Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 963 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.1 Content-Length: 304 v=0 o=201 1770785376 570897782 IN IP4 192.168.1.43 s=- c=IN IP4 192.168.1.43 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 --- (14 headers 14 lines) --- Using INVITE request as basis request - cjlqmdbgmjanzth@192.168.1.43 Sending to 192.168.1.43 : 5060 (NAT) Found user '201' Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.43:8000 Peer video RTP is at port 192.168.1.43:65535 Found description format speex Found description format speex Found description format PCMA Found description format PCMU Found description format GSM Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 74957757566 in office (domain 192.168.1.6) list_route: hop: Transmitting (no NAT) to 192.168.1.43:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.43;branch=z9hG4bKsyufaoni;received=192.168.1.43;rport=5060 From: "201" ;tag=isiyh To: Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 963 INVITE User-Agent: Usr Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- We're at 212.122.x.x port 12684 Video is at 212.122.x.x port 10784 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 212.53.40.40:5060: INVITE sip:74957757566@sipnet.ru SIP/2.0 Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK7aadd3b0;rport From: "SipPhone" ;tag=as3442dc46 To: Contact: Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 102 INVITE User-Agent: Usr Max-Forwards: 70 Date: Mon, 14 Apr 2008 13:50:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 218 v=0 o=root 18815 18815 IN IP4 212.122.x.x s=session c=IN IP4 212.122.x.x t=0 0 m=audio 12684 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- sip-gw*CLI> <-- SIP read from 212.53.40.40:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK7aadd3b0;rport=5060 From: "SipPhone" ;tag=as3442dc46 To: Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 102 INVITE Server: CommuniGatePro/5.2.2p Content-Length: 0 --- (8 headers 0 lines) --- <-- SIP read from 212.53.40.40:5060: SIP/2.0 401 Authentication required Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK7aadd3b0;rport=5060 From: "SipPhone" ;tag=as3442dc46 To: ;tag=9A4C65AB Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 102 INVITE WWW-Authenticate: Digest realm="etc.tario.ru",nonce="1E6BFA84146B441D1C62",opaque="opaqueData",qop="auth",algorithm=MD5 Server: CommuniGatePro/5.2.2p Content-Length: 0 --- (9 headers 0 lines) --- Transmitting (no NAT) to 212.53.40.40:5060: ACK sip:74957757566@sipnet.ru SIP/2.0 Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK7aadd3b0;rport From: "SipPhone" ;tag=as3442dc46 To: ;tag=9A4C65AB Contact: Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 102 ACK User-Agent: Usr Max-Forwards: 70 Content-Length: 0 --- We're at 212.122.x.x port 12684 Video is at 212.122.x.x port 10784 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 212.53.40.40:5060: NVITE sip:74957757566@sipnet.ru SIP/2.0 Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK774b3c23;rport From: "SipPhone" ;tag=as3442dc46 To: Contact: Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 103 INVITE User-Agent: Usr Max-Forwards: 70 Authorization: Digest username="user", realm="etc.tario.ru", algorithm=MD5, uri="sip:74957757566@sipnet.ru", nonce="1E6BFA84146B441D1C62", response="509c3ba604e4c4a5076900a151c3b753", opaque="opaqueData", qop=auth, cnonce="67a0e2f6", nc=00000001 Date: Mon, 14 Apr 2008 13:50:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 218 v=0 o=root 18815 18816 IN IP4 212.122.x.x s=session c=IN IP4 212.122.x.x t=0 0 m=audio 12684 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- sip-gw*CLI> <-- SIP read from 212.53.40.40:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK774b3c23;rport=5060 From: "SipPhone" ;tag=as3442dc46 To: Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 103 INVITE Server: CommuniGatePro/5.2.2p Content-Length: 0 --- (8 headers 0 lines) --- sip-gw*CLI> <-- SIP read from 212.53.40.40:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 212.122.x.x:5060;rport=5060;branch=z9hG4bK774b3c23 Record-Route: , Record-Route: Record-Route: From: ;tag=as3442dc46 To: ;tag=710a1ca5-136174 Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 103 INVITE Server: TarioSoftswitch/3.2.10 Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (no NAT) to 192.168.1.43:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.43;branch=z9hG4bKsyufaoni;received=192.168.1.43;rport=5060 From: "201" ;tag=isiyh To: ;tag=as589b0e88 Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 963 INVITE User-Agent: Usr Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- sip-gw*CLI> <-- SIP read from 212.53.40.40:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 212.122.x.x:5060;rport=5060;branch=z9hG4bK774b3c23 Record-Route: , Record-Route: Record-Route: From: ;tag=as3442dc46 To: ;tag=710a1ca5-136174 Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 103 INVITE Content-Type: application/sdp Server: TarioSoftswitch/3.2.10 Content-Length: 178 v=0 o=Tario-Softswitch 7115 101 IN IP4 212.53.40.82 s=SIP Call c=IN IP4 212.53.40.82 t=0 0 m=audio 21588 RTP/AVP 8 c=IN IP4 212.53.40.82 a=rtpmap:8 PCMA/8000 a=ptime:20 --- (12 headers 9 lines) --- Found RTP audio format 8 Peer audio RTP is at port 212.53.40.82:21588 Peer video RTP is at port 212.53.40.82:65535 Found description format PCMA Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) We're at 192.168.1.6 port 16014 Video is at 192.168.1.6 port 18442 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (no NAT) to 192.168.1.43:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.43;branch=z9hG4bKsyufaoni;received=192.168.1.43;rport=5060 From: "201" ;tag=isiyh To: ;tag=as589b0e88 Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 963 INVITE User-Agent: Usr Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18815 18815 IN IP4 192.168.1.6 s=session c=IN IP4 192.168.1.6 t=0 0 m=audio 16014 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- sip-gw*CLI> <-- SIP read from 212.53.40.40:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.122.x.x:5060;rport=5060;branch=z9hG4bK774b3c23 Record-Route: , Record-Route: Record-Route: From: ;tag=as3442dc46 To: ;tag=710a1ca5-136174 Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 103 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, BYE, CANCEL, INFO, OPTIONS Server: TarioSoftswitch/3.2.10 Content-Length: 178 v=0 o=Tario-Softswitch 7115 101 IN IP4 212.53.40.82 s=SIP Call c=IN IP4 212.53.40.82 t=0 0 m=audio 21588 RTP/AVP 8 c=IN IP4 212.53.40.82 a=rtpmap:8 PCMA/8000 a=ptime:20 --- (14 headers 9 lines) --- Found RTP audio format 8 Peer audio RTP is at port 212.53.40.82:21588 Peer video RTP is at port 212.53.40.82:65535 Found description format PCMA Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: list_route: hop: list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 212.53.40.40, port 5060 Transmitting (no NAT) to 212.53.40.40:5060: ACK sip:proc-136174@212.53.35.244 SIP/2.0 Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK13c47bee;rport Route: ,,, From: "SipPhone" ;tag=as3442dc46 To: ;tag=710a1ca5-136174 Contact: Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 103 ACK User-Agent: Usr Max-Forwards: 70 Content-Length: 0 --- We're at 192.168.1.6 port 16014 Video is at 192.168.1.6 port 18442 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.43:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.43;branch=z9hG4bKsyufaoni;received=192.168.1.43;rport=5060 From: "201" ;tag=isiyh To: ;tag=as589b0e88 Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 963 INVITE User-Agent: Usr Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 18815 18816 IN IP4 192.168.1.6 s=session c=IN IP4 192.168.1.6 t=0 0 m=audio 16014 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- set_destination: Parsing for address/port to send to set_destination: set destination to 212.53.40.40, port 5060 We're at 212.122.x.x port 12684 Video is at 212.122.x.x port 10784 Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x200 (speex) to SDP 14 headers, 11 lines sip-gw*CLI> <-- SIP read from 192.168.1.43:5060: ACK sip:74957757566@192.168.1.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.43;rport;branch=z9hG4bKiqivsnmf Max-Forwards: 70 Proxy-Authorization: Digest username="201",realm="asterisk",nonce="38433a2b",uri="sip:74957757566@192.168.1.6",response="a5f436e27fd3e6e3fb7044eb0fc485c6",algorithm=MD5 To: ;tag=as589b0e88 From: "201" ;tag=isiyh Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 963 ACK User-Agent: Twinkle/1.1 Content-Length: 0 --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.43, port 5060 We're at 192.168.1.6 port 16014 Video is at 192.168.1.6 port 18442 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.1.43:5060: INVITE sip:201@192.168.1.43 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK72d35c20;rport From: ;tag=as589b0e88 To: "201" ;tag=isiyh Contact: Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 102 INVITE User-Agent: Usr Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 216 v=0 o=root 18815 18817 IN IP4 212.53.40.82 s=session c=IN IP4 212.53.40.82 t=0 0 m=audio 21588 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- sip-gw*CLI> <-- SIP read from 192.168.1.43:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;branch=z9hG4bK72d35c20 To: "201" ;tag=isiyh From: ;tag=as589b0e88 Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 102 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Server: Twinkle/1.1 Supported: replaces,norefersub Content-Length: 202 v=0 o=201 1770785376 570897783 IN IP4 192.168.1.43 s=- c=IN IP4 192.168.1.43 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 --- (12 headers 10 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.43:8000 Peer video RTP is at port 192.168.1.43:65535 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.43, port 5060 Transmitting (no NAT) to 192.168.1.43:5060: ACK sip:201@192.168.1.43 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK7af62bf0;rport From: ;tag=as589b0e88 To: "201" ;tag=isiyh Contact: Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 102 ACK User-Agent: Usr Max-Forwards: 70 Content-Length: 0 --- sip-gw*CLI> <-- SIP read from 212.53.40.40:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK1db4436c;rport=5060 From: "SipPhone" ;tag=as3442dc46 To: ;tag=710a1ca5-136174 Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 104 INVITE Server: CommuniGatePro/5.2.2p Content-Length: 0 --- (8 headers 0 lines) --- sip-gw*CLI> <-- SIP read from 212.53.40.40:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 212.122.x.x:5060;rport=5060;branch=z9hG4bK1db4436c From: ;tag=as3442dc46 To: ;tag=710a1ca5-136174 Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 104 INVITE Server: TarioSoftswitch/3.2.10 Content-Length: 0 --- (8 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 212.53.40.40, port 5060 Transmitting (no NAT) to 212.53.40.40:5060: ACK sip:proc-136174@212.53.35.244 SIP/2.0 Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK1db4436c;rport Route: ,,, From: "SipPhone" ;tag=as3442dc46 To: ;tag=710a1ca5-136174 Contact: Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 104 ACK User-Agent: Usr Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call 'cjlqmdbgmjanzth@192.168.1.43' in 32000 ms set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.43, port 5060 Reliably Transmitting (no NAT) to 192.168.1.43:5060: BYE sip:201@192.168.1.43 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK27a3b839;rport From: ;tag=as589b0e88 To: "201" ;tag=isiyh Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 103 BYE User-Agent: Usr Max-Forwards: 70 Content-Length: 0 --- sip-gw*CLI> <-- SIP read from 192.168.1.43:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;branch=z9hG4bK27a3b839 To: "201" ;tag=isiyh From: ;tag=as589b0e88 Call-ID: cjlqmdbgmjanzth@192.168.1.43 CSeq: 103 BYE Server: Twinkle/1.1 Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru' Destroying call 'cjlqmdbgmjanzth@192.168.1.43' sip-gw*CLI> ~~~~~~~~~~~~~~~ No such command '~~~~~~~~~~~~~~~' (type 'help' for help) sip-gw*CLI> sip-gw*CLI> sip-gw*CLI> sip-gw*CLI> sip-gw*CLI> sip-gw*CLI> sip-gw*CLI> sip-gw*CLI> sip-gw*CLI> sip-gw*CLI> sip-gw*CLI> Apr 15 00:50:49 NOTICE[18815]: chan_sip.c:5460 sip_reregister: -- Re-registration for user@sipnet.ru REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 212.53.40.40:5060: REGISTER sip:sipnet.ru SIP/2.0 Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK00df1a32;rport From: ;tag=as6e0e9def To: Call-ID: 3fa188a21eb4b93353d294f07baaba69@192.168.1.43 CSeq: 116 REGISTER User-Agent: Usr Max-Forwards: 70 Authorization: Digest username="user", realm="etc.tario.ru", algorithm=MD5, uri="sip:sipnet.ru", nonce="1E6BFA84146B441D1C62", response="c7e34d3459edae65d95fb78737b29dfb", opaque="opaqueData", qop=auth, cnonce="521aa385", nc=00000002 Expires: 120 Contact: Event: registration Content-Length: 0 --- sip-gw*CLI> <-- SIP read from 212.53.40.40:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK00df1a32;rport=5060 From: ;tag=as6e0e9def To: ;tag=636BD494 Call-ID: 3fa188a21eb4b93353d294f07baaba69@192.168.1.6 CSeq: 116 REGISTER Expires: 120 Contact: ;expires=120 Event: registration Date: Mon, 14 Apr 2008 13:50:49 GMT Allow: PUBLISH,SUBSCRIBE Supported: path Allow-Events: presence,message-summary,reg,dialog,line-seize,keep-alive,refer Server: CommuniGatePro/5.2.2p Content-Length: 0 --- (15 headers 0 lines) --- Scheduling destruction of call '3fa188a21eb4b93353d294f07baaba69@192.168.1.6' in 32000 ms Apr 15 00:50:49 NOTICE[18815]: chan_sip.c:9979 handle_response_register: Outbound Registration: Expiry for sipnet.ru is 120 sec (Scheduling reregistration in 105 s) Destroying call '3fa188a21eb4b93353d294f07baaba69@192.168.1.6' sip-gw*CLI> <-- SIP read from 212.53.40.40:5060: BYE sip:user@212.122.x.x SIP/2.0 Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK741122-lnqxjfj;cgp=etc.tario.ru;upaddr=212.53.35.244;rport Via: SIP/2.0/UDP 212.53.35.244;branch=z9hG4bK2014814026 Via: SIP/2.0/TCP 212.53.35.244:50666 Max-Forwards: 68 From: ;tag=710a1ca5-136174 To: ;tag=as3442dc46 Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 53496 BYE User-Agent: TarioSoftswitch/3.2.10 Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (no NAT) to 212.53.40.40:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK741122-lnqxjfj;cgp=etc.tario.ru;upaddr=212.53.35.244;received=212.53.40.40;rport=5060 Via: SIP/2.0/UDP 212.53.35.244;branch=z9hG4bK2014814026 Via: SIP/2.0/TCP 212.53.35.244:50666 From: ;tag=710a1ca5-136174 To: ;tag=as3442dc46 Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru CSeq: 53496 BYE User-Agent: Usr Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Apr 15 00:52:34 NOTICE[18815]: chan_sip.c:5460 sip_reregister: -- Re-registration for user@sipnet.ru REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 212.53.40.40:5060: REGISTER sip:sipnet.ru SIP/2.0 Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK1245854e;rport From: ;tag=as7c66f301 To: Call-ID: 3fa188a21eb4b93353d294f07baaba69@192.168.1.6 CSeq: 117 REGISTER User-Agent: Usr Max-Forwards: 70 Authorization: Digest username="user", realm="etc.tario.ru", algorithm=MD5, uri="sip:sipnet.ru", nonce="1E6BFA84146B441D1C62", response="2b7cc1f06931754357723c2255d320b7", opaque="opaqueData", qop=auth, cnonce="50e65be7", nc=00000003 Expires: 120 Contact: Event: registration Content-Length: 0 --- sip-gw*CLI> <-- SIP read from 212.53.40.40:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK1245854e;rport=5060 From: ;tag=as7c66f301 To: ;tag=07737923 Call-ID: 3fa188a21eb4b93353d294f07baaba69@192.168.1.6 CSeq: 117 REGISTER Expires: 120 Contact: ;expires=120 Event: registration Date: Mon, 14 Apr 2008 13:52:34 GMT Allow: PUBLISH,SUBSCRIBE Supported: path Allow-Events: presence,message-summary,reg,dialog,line-seize,keep-alive,refer Server: CommuniGatePro/5.2.2p Content-Length: 0 --- (15 headers 0 lines) --- Scheduling destruction of call '3fa188a21eb4b93353d294f07baaba69@192.168.1.6' in 32000 ms Apr 15 00:52:34 NOTICE[18815]: chan_sip.c:9979 handle_response_register: Outbound Registration: Expiry for sipnet.ru is 120 sec (Scheduling reregistration in 105 s) Destroying call '3fa188a21eb4b93353d294f07baaba69@192.168.1.6'