Hi List, > Try canreinvite=yes in sip trunk
This did not make any difference... -.- > > -----Original Message----- > > Hi @ll, > > I just moved my Asterisk Box and changed the Provider and Internet Access > to a full IP Access by Deutsche Telekom. > > I set up my sip.conf as I found various examples throughout the Net. Calls > and some other stuff is basically working. > > The problem I ran into is, that the outgoing and incoming calls are > dropped after exactly 15 Minutes. Solution for this should be setting the > session-timers to refuse but this doesnt change anything here. > > I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest > Asterisk by Digium without success. > > Has anyone else has the Same problem or is a solution already known? Could > someone point me in the right direction? I can provide (debug) logs if > essential. > > Best regards > > Flo > > -- _____________________________________________________________________ -- 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
