[Asterisk-Users] Sip call hang up

[Eduardo Goncalves]

Hi list,

	I have a cisco 827 with 4 fxs and an * gateway, like this:

[c827]------sip-----[asterisk]-----e&m---PSTN

	The codec used is g711alaw over a 9Mb link.
	Some calls just hang up after some minutes of conversation. Cisco shows
a  "DisconnectText=normal call clearing (16)" and I found nothing in
asterisk logs.
	Anyone can help?

thanks
Eduardo