This is a classic symptom of having reinvites and/or direct media enabled on Asterisk or SIP ALG enabled on the router.
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Cameo Sent: Monday, January 06, 2014 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped call on new CISCO router for no reason! Hello Everyone, Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT setup. After 2 seconds on a successfully established call we reach retrans max, and asterisk disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as expected. Your help is greatly appreciated, Nick. -- _____________________________________________________________________ -- 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
