Package: asterisk Version: 1:1.6.2.0~dfsg~rc1-1 Hello,
with this version of asterisk, the chan_sip modules does not free rtp sockets anymore under certain cirumstances. I have not been able to trigger this behaviour on purpose, however. When this version runs for a while (a few days) and handles a few dozen calls, the whole rtp socket range is used up not still held open by asterisk, although all calls have ended. In this state, it does not accept new sip calls, neither incoming nor outgoing. Bye, Joerg
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