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<TITLE>RE: [Asterisk-Dev] SIP Hard Disconnect Detection</TITLE>
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<P><FONT SIZE=2>But the problem is that I have my own new developed application (deployed the same way as for example "Playback" application) and even if I set rtptimeout and rtpholdtimeout to any values I don't know how to detect the channel was disconnected. My source code has a while loop and the channel isn't automatically killed. Can anyone help??</FONT></P>
<P><FONT SIZE=2>Thank you,</FONT>
<BR><FONT SIZE=2>Pedro Goncalves</FONT>
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<P><FONT SIZE=2>________________________________________</FONT>
<BR><FONT SIZE=2>From: asterisk-dev-admin@lists.digium.com [<A HREF="mailto:asterisk-dev-admin@lists.digium.com">mailto:asterisk-dev-admin@lists.digium.com</A>] On Behalf Of Pedro Bessa Goncalves</FONT></P>
<P><FONT SIZE=2>Sent: quarta-feira, 21 de Julho de 2004 19:45</FONT>
<BR><FONT SIZE=2>To: Asterisk-Dev@Lists. Digium. Com (asterisk-dev@lists.digium.com); Asterisk-Users@Lists. Digium. Com (asterisk-users@lists.digium.com)</FONT></P>
<P><FONT SIZE=2>Subject: [Asterisk-Dev] SIP Hard Disconnect Detection</FONT>
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<P><FONT SIZE=2>Hello. I have a question regarding Asterisk internal API. </FONT>
<BR><FONT SIZE=2>I am developing a new asterisk module application using asterisk internal c API. I am having problem detecting hard hangups when the SIP clients disconnect (suppose power goes off in the phones). I am not receiving any disconnect control frames and don't know how to check if the clients are really connected. Can anyone help?</FONT></P>
<P><FONT SIZE=2>Thank you, </FONT>
<BR><FONT SIZE=2>Pedro Goncalves </FONT>
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