Try canreinvite=yes in sip trunk -----Original Message----- From: Florian Wolters <flor...@florian-wolters.de> Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 21 Mar 2013 08:31:54 To: <asterisk-users@lists.digium.com> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes
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 -- _____________________________________________________________________ -- 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