<-- SIP read from 202.171.182.120:5060: INVITE sip:97131806@203.82.162.33:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.51:5060;rport;branch=z9hG4bK28f8d27000084 From: 97131800;tag=8c113248 To: Call-ID: b7242eeb-13@192.168.1.51 CSeq: 248108771 INVITE Max-Forwards: 70 Contact: 97131800 User-Agent: Soundwin_ATA Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, SUBSCRIBE, REFER, NOTIFY, PRACK, UPDATE, MESSAGE, SERVICE, INFO Allow-Events: refer Supported: 100rel, replaces Content-Length: 309 Content-Type: application/sdp v=0 o=- 1369133069 1125898167 IN IP4 192.168.1.51 s=Voice call e=you@yourcompany.com c=IN IP4 192.168.1.51 t=0 0 a=sendrecv m=image 5000 udptl t38 a=sendrecv a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:280 a=T38FaxMaxDatagram:316 a=T38FaxUdpEC:t38UDPRedundancy --- (14 headers 14 lines)--- Sending to 202.171.182.120 : 5060 (NAT) Using INVITE request as basis request - b7242eeb-13@192.168.1.51 Reliably Transmitting (NAT) to 202.171.182.120:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK28f8d27000084;received=202.171.182.120;rport=5060 From: 97131800;tag=8c113248 To: ;tag=as0b37710d Call-ID: b7242eeb-13@192.168.1.51 CSeq: 248108771 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f417525" Content-Length: 0 --- Scheduling destruction of SIP dialog 'b7242eeb-13@192.168.1.51' in 32000 ms (Method: INVITE) Found user '97131800' thirtythree*CLI> <-- SIP read from 202.171.182.120:5060: ACK sip:97131806@203.82.162.33:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.51:5060;rport;branch=z9hG4bK28f8d27000084 From: 97131800;tag=8c113248 To: ;tag=as0b37710d Call-ID: b7242eeb-13@192.168.1.51 CSeq: 248108771 ACK Max-Forwards: 70 User-Agent: Soundwin_ATA Content-Length: 0 --- (9 headers 0 lines)--- thirtythree*CLI> <-- SIP read from 202.171.182.120:5060: INVITE sip:97131806@203.82.162.33:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.51:5060;rport;branch=z9hG4bK1396c49000085 From: 97131800;tag=8c113248 To: Call-ID: b7242eeb-13@192.168.1.51 CSeq: 248108772 INVITE Max-Forwards: 70 Contact: 97131800 User-Agent: Soundwin_ATA Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, SUBSCRIBE, REFER, NOTIFY, PRACK, UPDATE, MESSAGE, SERVICE, INFO Allow-Events: refer Supported: 100rel, replaces Proxy-Authorization: Digest username="97131800", realm="asterisk", uri="sip:97131806@203.82.162.33", nonce="4f417525", response="895606bdec8a11eb0d84e112ec4f1c92" Content-Length: 309 Content-Type: application/sdp v=0 o=- 1369133069 1125898167 IN IP4 192.168.1.51 s=Voice call e=you@yourcompany.com c=IN IP4 192.168.1.51 t=0 0 a=sendrecv m=image 5000 udptl t38 a=sendrecv a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:280 a=T38FaxMaxDatagram:316 a=T38FaxUdpEC:t38UDPRedundancy --- (15 headers 14 lines)--- Sending to 202.171.182.120 : 5060 (NAT) Using INVITE request as basis request - b7242eeb-13@192.168.1.51 Found user '97131800' Got T.38 offer in SDP in dialog b7242eeb-13@192.168.1.51 Peer doesn't provide audio Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x3d0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|h261|h263|h263p|h264)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x7 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.51:5000 Peer video RTP is at port 192.168.1.51:24585 Looking for 97131806 in from-internal (domain 203.82.162.33) list_route: hop: Transmitting (NAT) to 202.171.182.120:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK1396c49000085;received=202.171.182.120;rport=5060 From: 97131800;tag=8c113248 To: Call-ID: b7242eeb-13@192.168.1.51 CSeq: 248108772 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- Audio is at 203.82.162.33 port 11354 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 202.171.182.120:10800: INVITE sip:97131806@192.168.1.54:5060 SIP/2.0 Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK4883837d;rport From: "97131800" ;tag=as7cc2ac54 To: Contact: Call-ID: 107dcb9d747564104f7a1dbf1aac0926@203.82.162.33 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 24 Jul 2006 02:46:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 254 v=0 o=root 19738 19738 IN IP4 203.82.162.33 s=session c=IN IP4 203.82.162.33 t=0 0 m=audio 11354 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- thirtythree*CLI> <-- SIP read from 202.171.182.120:10800: SIP/2.0 100 Trying Via: SIP/2.0/UDP 203.82.162.33:5060;rport=5060;received=203.82.162.33;branch=z9hG4bK4883837d From: "97131800";tag=as7cc2ac54 To: Call-ID: 107dcb9d747564104f7a1dbf1aac0926@203.82.162.33 CSeq: 102 INVITE Max-Forwards: 70 User-Agent: Soundwin_ATA Content-Length: 0 --- (9 headers 0 lines)--- thirtythree*CLI> <-- SIP read from 202.171.182.120:10800: SIP/2.0 488 Incompatible Media Format Via: SIP/2.0/UDP 203.82.162.33:5060;rport=5060;received=203.82.162.33;branch=z9hG4bK4883837d From: "97131800";tag=as7cc2ac54 To: ;tag=15a4583 Call-ID: 107dcb9d747564104f7a1dbf1aac0926@203.82.162.33 CSeq: 102 INVITE Max-Forwards: 70 User-Agent: Soundwin_ATA Warning: 305 ATA.Soundwin "Incompatible Media Format" Content-Length: 0 --- (10 headers 0 lines)--- Transmitting (NAT) to 202.171.182.120:10800: ACK sip:97131806@192.168.1.54:5060 SIP/2.0 Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK4883837d;rport From: "97131800" ;tag=as7cc2ac54 To: ;tag=15a4583 Contact: Call-ID: 107dcb9d747564104f7a1dbf1aac0926@203.82.162.33 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Transmitting (NAT) to 202.171.182.120:5060: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK1396c49000085;received=202.171.182.120;rport=5060 From: 97131800;tag=8c113248 To: ;tag=as716bcd36 Call-ID: b7242eeb-13@192.168.1.51 CSeq: 248108772 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 --- thirtythree*CLI> <-- SIP read from 202.171.182.120:5060: ACK sip:97131806@203.82.162.33:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.51:5060;rport;branch=z9hG4bK1396c49000085 From: 97131800;tag=8c113248 To: ;tag=as716bcd36 Call-ID: b7242eeb-13@192.168.1.51 CSeq: 248108772 ACK Max-Forwards: 70 User-Agent: Soundwin_ATA Content-Length: 0 --- (9 headers 0 lines)--- Really destroying SIP dialog '107dcb9d747564104f7a1dbf1aac0926@203.82.162.33' Me Really destroying SIP dialog 'b7242eeb-13@192.168.1.51' Method: ACK