On Tuesday 25 March 2008 10:17:54 Vieri wrote: > --- Steve Davies <[EMAIL PROTECTED]> wrote: > > Using rtptimeout and rtpholdtimeout settings in > > sip.conf > > I set > rtptimeout=10 > rtpholdtimeout=30 > (just for testing; I know these values are way too > low) > then did a > CLI> sip reload > and waited more than 30 seconds. > > The SIP channel is still there (InUse). > Channel Location State > Application(Data) > SIP/6010-b38d53e0 [EMAIL PROTECTED]:8 Up > Dial(SIP/4053||tTwW) > > Should I interpret the above that it's in an infinite > loop trying to dial/reach SIP/4053?
Given that you didn't give Dial a timeout, yes, it will try forever, until it receives a response. Note that this has nothing to do with rtptimeout, as that takes effect when the call is established, and the RTP packets stop flowing. Without using a firewall rule between the two hosts, it is somewhat difficult to mock up that situation (as the RTP is still flowing, even if the audio is silent). -- Tilghman _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
