sol*CLI> <-- SIP read from 24.76.187.116:1414: INVITE sip:7916118@sol2.bighostbox.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-f2576c2d From: ;tag=f04cb8d053c6f08co0 To: Remote-Party-ID: ;screen=yes;party=calling Call-ID: d9754120-8b17847c@192.168.1.130 CSeq: 101 INVITE Max-Forwards: 70 Contact: Expires: 240 User-Agent: Sipura/SPA941-4.1.8 Content-Length: 401 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE Allow-Events: dialog Content-Type: application/sdp v=0 o=- 58433693 58433693 IN IP4 192.168.1.130 s=- c=IN IP4 192.168.1.130 t=0 0 m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (15 headers 18 lines)--- Using INVITE request as basis request - d9754120-8b17847c@192.168.1.130 Sending to 192.168.1.130 : 5060 (NAT) Reliably Transmitting (NAT) to 24.76.187.116:1414: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-f2576c2d;received=24.76.187.116 From: ;tag=f04cb8d053c6f08co0 To: ;tag=as08979d78 Call-ID: d9754120-8b17847c@192.168.1.130 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="0185d33e" Content-Length: 0 --- Scheduling destruction of call 'd9754120-8b17847c@192.168.1.130' in 15000 ms Found user '2048850872x3' sol*CLI> <-- SIP read from 24.76.187.116:1414: ACK sip:7916118@sol2.bighostbox.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-f2576c2d From: ;tag=f04cb8d053c6f08co0 To: ;tag=as08979d78 Call-ID: d9754120-8b17847c@192.168.1.130 CSeq: 101 ACK Max-Forwards: 70 Contact: User-Agent: Sipura/SPA941-4.1.8 Content-Length: 0 Allow-Events: dialog --- (11 headers 0 lines)--- sol*CLI> <-- SIP read from 24.76.187.116:1414: INVITE sip:7916118@sol2.bighostbox.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-7e82041d From: ;tag=f04cb8d053c6f08co0 To: Remote-Party-ID: ;screen=yes;party=calling Call-ID: d9754120-8b17847c@192.168.1.130 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="2048850872x3",realm="asterisk",nonce="0185d33e",uri="sip:7916118@sol2.bighostbox.com",algorithm=MD5,response="30f75f02ff0f4648bdecc6f75ebabf23" Contact: Expires: 240 User-Agent: Sipura/SPA941-4.1.8 Content-Length: 401 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE Allow-Events: dialog Content-Type: application/sdp v=0 o=- 58433693 58433693 IN IP4 192.168.1.130 s=- c=IN IP4 192.168.1.130 t=0 0 m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (16 headers 18 lines)--- Using INVITE request as basis request - d9754120-8b17847c@192.168.1.130 Sending to 192.168.1.130 : 5060 (NAT) Found user '2048850872x3' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.130:16450 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format telephone-event Capabilities: us - 0x516 (gsm|ulaw|g726|g729|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x514 (ulaw|g726|g729|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 7916118 in openit (domain sol2.bighostbox.com) list_route: hop: Transmitting (NAT) to 24.76.187.116:1414: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-7e82041d;received=24.76.187.116 From: ;tag=f04cb8d053c6f08co0 To: Call-ID: d9754120-8b17847c@192.168.1.130 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- We're at 64.201.170.160 port 13712 Adding codec 0x10 (g726) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (NAT) to 24.76.187.116:1414: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-7e82041d;received=24.76.187.116 From: ;tag=f04cb8d053c6f08co0 To: ;tag=as4d1aa95f Call-ID: d9754120-8b17847c@192.168.1.130 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 341 v=0 o=root 6515 6515 IN IP4 64.201.170.160 s=session c=IN IP4 64.201.170.160 t=0 0 m=audio 13712 RTP/AVP 2 3 0 97 18 101 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Transmitting (NAT) to 24.76.187.116:1414: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-7e82041d;received=24.76.187.116 From: ;tag=f04cb8d053c6f08co0 To: ;tag=as4d1aa95f Call-ID: d9754120-8b17847c@192.168.1.130 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- sol*CLI> <-- SIP read from 24.76.187.116:1414: CANCEL sip:7916118@sol2.bighostbox.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-7e82041d From: ;tag=f04cb8d053c6f08co0 To: Call-ID: d9754120-8b17847c@192.168.1.130 CSeq: 102 CANCEL Max-Forwards: 70 Proxy-Authorization: Digest username="2048850872x3",realm="asterisk",nonce="0185d33e",uri="sip:7916118@sol2.bighostbox.com",algorithm=MD5,response="d7424d116fb1a111a39bd2f9cfb8b051" User-Agent: Sipura/SPA941-4.1.8 Content-Length: 0 --- (10 headers 0 lines)--- Sending to 192.168.1.130 : 5060 (NAT) Reliably Transmitting (NAT) to 24.76.187.116:1414: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-7e82041d;received=24.76.187.116 From: ;tag=f04cb8d053c6f08co0 To: ;tag=as4d1aa95f Call-ID: d9754120-8b17847c@192.168.1.130 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 24.76.187.116:1414: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-7e82041d;received=24.76.187.116 From: ;tag=f04cb8d053c6f08co0 To: ;tag=as4d1aa95f Call-ID: d9754120-8b17847c@192.168.1.130 CSeq: 102 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- sol*CLI> <-- SIP read from 24.76.187.116:1414: ACK sip:7916118@sol2.bighostbox.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK-7e82041d From: ;tag=f04cb8d053c6f08co0 To: ;tag=as4d1aa95f Call-ID: d9754120-8b17847c@192.168.1.130 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="2048850872x3",realm="asterisk",nonce="0185d33e",uri="sip:7916118@sol2.bighostbox.com",algorithm=MD5,response="30d71eb1a8b9a568102f46f7659e4c96" Contact: User-Agent: Sipura/SPA941-4.1.8 Content-Length: 0 Allow-Events: dialog --- (12 headers 0 lines)--- Destroying call 'd9754120-8b17847c@192.168.1.130'