<div><div><br></div><div>--- (12 headers 0 lines) ---</div><div>Sending to 192.168.0.50 : 12714 (NAT)</div><div>Transmitting (NAT) to <a href="http://192.168.0.50:12714">192.168.0.50:12714</a>:</div><div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP 192.168.0.50:12714;branch=z9hG4bK-d87543-930550325154e53d-1--d87543-;received=192.168.0.50;rport=12714</div><div>From: "cc106"<<a href="mailto:sip%3Acc106@192.168.0.2">sip:cc106@192.168.0.2</a>>;tag=7f1cff22</div>
<div>To: "817275691533"<<a href="mailto:sip%3A817275691533@192.168.0.2">sip:817275691533@192.168.0.2</a>>;tag=as02559696</div><div>Call-ID: NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.</div><div>CSeq: 3 BYE</div>
<div>User-Agent: Asterisk PBX</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div><div>Contact: <<a href="mailto:sip%3A817275691533@192.168.0.2">sip:817275691533@192.168.0.2</a>></div><div>
Content-Length: 0</div><div>X-Asterisk-HangupCause: Normal Clearing</div><div><br></div><div><br></div><div><br></div><div>---</div><div>Scheduling destruction of call '617ad67d47db8e4a2155fcd51d1089ff@59.xxx.xx.xx' in 32000 ms</div>
<div>set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for address/port to send to</div><div>set_destination: set destination to 8.14.xxx.xxx, port 5060</div><div>Reliably Transmitting (no NAT) to 8.14.xxx.xxx:5060:</div>
<div>BYE sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0</div><div>Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK59c0212a;rport</div><div>From: "cc106" <sip:fiddialer@59.xxx.xx.xx>;tag=as3f9466a7</div><div>
To: <sip:17275691533@8.14.xxx.xxx>;tag=1902000923108720995156225</div><div>Call-ID: 617ad67d47db8e4a2155fcd51d1089ff@59.xxx.xx.xx</div><div>CSeq: 103 BYE</div><div>User-Agent: Asterisk PBX</div><div>Max-Forwards: 70</div>
<div>Content-Length: 0</div><div><br></div><div><br></div><div>---</div><div> == Spawn extension (default, 817275691533, 2) exited non-zero on 'SIP/cc106-b7a1a9d0'</div><div> -- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi://<a href="http://127.0.0.1:4577/call_log">127.0.0.1:4577/call_log</a>") in new stack</div>
<div> -- AGI Script agi://<a href="http://127.0.0.1:4577/call_log">127.0.0.1:4577/call_log</a> completed, returning 0</div><div> -- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi://<a href="http://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12">127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12</a>)") in new stack</div>
<div> -- AGI Script agi://<a href="http://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12">127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12</a>) completed, returning 0</div>
<div>Destroying call 'NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.'</div><div>vicidialnow*CLI></div><div><-- SIP read from 8.14.xxx.xxx:5060:</div><div>SIP/2.0 200 OK</div><div>CSeq: 102 INVITE</div><div>Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK3a111ef4;rport</div>
<div>From: "V0219160007000134649" <sip:fiddialer@59.xxx.xx.xx>;tag=as79fae976</div><div>Call-ID: 1525f0ef1e787bed51ed1ef119adb1fa@59.xxx.xx.xx</div><div>To: <sip:16785588539@8.14.xxx.xxx>;tag=1902000923098720982816221</div>
<div>Contact: <sip:8.14.xxx.xxx:5060;transport=udp></div><div>Content-Type: application/sdp</div><div>Content-Length: 225</div><div><br></div><div>v=0</div><div>o=VoipSwitch 7220 7220 IN IP4 8.14.xxx.xxx</div><div>s=VoipSIP</div>
<div>i=Audio Session</div><div>c=IN IP4 8.14.xxx.xxx</div><div>t=0 0</div><div>m=audio 6220 RTP/AVP 18 101</div><div>a=rtpmap:18 G729/8000/1</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-15</div><div>
a=sendrecv</div><div><br></div><div>--- (9 headers 11 lines) ---</div><div>Found RTP audio format 18</div><div>Found RTP audio format 101</div><div>Peer audio RTP is at port 8.14.xxx.xxx:6220</div><div>Found description format G729</div>
<div>Found description format telephone-event</div><div>Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)</div><div>Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)</div>
<div>list_route: hop: <sip:8.14.xxx.xxx:5060;transport=udp></div><div>set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for address/port to send to</div><div>set_destination: set destination to 8.14.xxx.xxx, port 5060</div>
<div>Transmitting (no NAT) to 8.14.xxx.xxx:5060:</div><div>ACK sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0</div><div>Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK6eef7893;rport</div><div>From: "V0219160007000134649" <sip:fiddialer@59.xxx.xx.xx>;tag=as79fae976</div>
<div>To: <sip:16785588539@8.14.xxx.xxx>;tag=1902000923098720982816221</div><div>Contact: <sip:fiddialer@59.xxx.xx.xx></div><div>Call-ID: 1525f0ef1e787bed51ed1ef119adb1fa@59.xxx.xx.xx</div><div>CSeq: 102 ACK</div>
<div>User-Agent: Asterisk PBX</div><div>Max-Forwards: 70</div><div>Content-Length: 0</div><div><br></div><div><br></div></div>