[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[isrlgb at gmail.com]

Try canreinvite=yes in sip trunk

-----Original Message-----
From: Florian Wolters <florian at florian-wolters.de>
Sender: asterisk-users-bounces at lists.digium.com
Date: Thu, 21 Mar 2013 08:31:54 
To: <asterisk-users at lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at 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