[Asterisk-Dev] SIP Hard Disconnect Detection

[Pedro Bessa Goncalves]

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??

Thank you,
Pedro Goncalves

________________________________________
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] On Behalf Of Pedro Bessa
Goncalves
Sent: quarta-feira, 21 de Julho de 2004 19:45
To: Asterisk-Dev at Lists. Digium. Com (asterisk-dev at lists.digium.com);
Asterisk-Users at Lists. Digium. Com (asterisk-users at lists.digium.com)
Subject: [Asterisk-Dev] SIP Hard Disconnect Detection

Hello. I have a question regarding Asterisk internal API. 
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?
Thank you, 
Pedro Goncalves 
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