>> > I don't think that is the case at all. I believe that all calls > will carry on without interruption. > > Julian
Well I learned something new today... thank you. In the past, I could swear that reloading chan_sip on a bridged call would cause me to loose connection. But now, I cannot get even the reload to do this. Good to know. I am running Asterisk SVN-branch-1.4-r89125M and this does not occur. Thanks again Julian. _______________________________________________ --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
