On Fri, Feb 21, 2014 at 10:55 AM, Ruddy Gbaguidi <[email protected]> wrote: > Hi, > Here is my scenario. > I have a SIP call between two SIP endpoints. A calls B. > During the ringing, B disconnects (network cable is unplugged). > > But A continue ringing forever (until the dial timeout) even if asterisk > detects that B is disconnected with the qualify. > > Is there any setup or asterisk configuration I need to enable to have A > close its call ? > > Note: when A is already talking with B, the call is hanged up on rtp > timeout. But not during the Ringing phase.
I'm not sure it is possible to configure Asterisk to hang up during the ringing phase when a peer/endpoint becomes unreachable. I don't see an option or parameter for that behavior. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
