On 25/03/2008, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > 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).
I would have assumed that "Up" indicates that it thinks the call is already in-progress, and that RTP should be flowing. You will not be able to change the rtptimeout values of an established call as AFAIK these settings are copied into the channel object when it is created, and not updated if SIP is reloaded. So... I think that channel is stuck until a restart, but you may be able to reduce its occurrence in future by using rtptimeout, and possibly also by using Dial timeouts as Tilghman pointed out. I use rtptimeout as standard these days, and have not seen a channel stuck-hard for quite a while (Nor have I seen false hang-ups). Regards, Steve _______________________________________________ -- 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
