On Tue, Nov 09, 2010 at 01:50:23PM +0100, Joerg Dorchain wrote: > On Mon, Nov 08, 2010 at 08:04:32PM -0500, Paul Belanger wrote: > > > > Please retest with the latest of Asterisk (1:1.6.2.9-2). Some recent > > upstream bugs were resolved regarding chan_sip and sockets. > > At first glance it looks better. The first call with > session-timers=originate did not leave open sockets. > I'll keep phoning...
And so I did. In short: Negativ. Asterisk keeps rtp sockets open. I have the impression it happens more quickly when calls are transfered and then "bridged nativly". Maybe asterisk forgets its rtp sockets in this situation. The workaround session-timers = refuse in sip.conf is still necessary. Bye, Joerg
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