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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link="#0563C1" vlink="#954F72"><div class=WordSection1><p class=MsoNormal>I know this was discussed years ago – but I’m looking into whether things have changed. Imagine this scenario:<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l1 level1 lfo1'><![if !supportLists]><span style='mso-list:Ignore'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><![endif]>Phone A call Phone B through Asterisk. (A -- > Asterisk -- > B)<o:p></o:p></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l1 level1 lfo1'><![if !supportLists]><span style='mso-list:Ignore'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><![endif]>All 3 devices have public IP addresses, and Asterisk is configured for directmedia / reinvites.<o:p></o:p></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l1 level1 lfo1'><![if !supportLists]><span style='mso-list:Ignore'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><![endif]>Phone A and B are having a successful call with direct RTP.<o:p></o:p></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l1 level1 lfo1'><![if !supportLists]><span style='mso-list:Ignore'>4.<span style='font:7.0pt "Times New Roman"'> </span></span><![endif]>Asterisk shutdowns down (pull the power) and the SIP connection closes (maybe a FIN is sent, maybe not)<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>My questions are:<o:p></o:p></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo2'><![if !supportLists]><span style='mso-list:Ignore'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><![endif]>Will he call drop? <o:p></o:p></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo2'><![if !supportLists]><span style='mso-list:Ignore'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><![endif]>Immediately or after some SIP packet times out?<o:p></o:p></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo2'><![if !supportLists]><span style='mso-list:Ignore'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><![endif]>Is there a way to keep the call up without Asterisk/SIP? (This was discussed before and the practical answer was no) <o:p></o:p></p><p class=MsoListParagraph><o:p> </o:p></p><p class=MsoNormal>I’m curious if anything has changed. The only solution put forward years ago was adding a proxy in front of Asterisk which redirects SIP between phone, but that discussion had lots of negatives / debate.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p></div></body></html>