Thanks, I found solution - in Asterisk - rtptimeout I am using rtpproxy with Kamailio, there is timeout socket, but for this moment, it looks complicated for me.
On Mon, Feb 13, 2012 at 5:53 PM, David <[email protected]> wrote: > Doubt it. > > You would need your "media" gateway to detect such a case. > > > > > On 2/13/12 10:44 AM, Stoyan Mihaylov wrote: > > Problem - if connection drop, call can persist. > In Asterisk there is silencedetecthangup - which should cause hangup, if > there is full silence for desired period of time. > Unfortunately it does not hangup. > I mean: > SIP client 1 ->Kamailio -> Asterisk ->Kamailio -> SIP client 2 > If I drop connection for one of SIP clients, I expect call should be > automatically hangup after a time I set (20 sec). > But call persists. In worst case - if connection drops for both clients, > call will persist until Asterisk is restarted. > I will continue to look how to solve problem with Asterisk, but I am > curious if this can be done from Kamailio also. > If I can cancel call from both places - it will be great. > I need to ensure that if something wrong happens, call will be dropped > within 30 sec maximum. > Stoyan > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > [email protected]http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > [email protected] > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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