[asterisk-users] Dropped call on new CISCO router for no reason!

[Eric Wieling]

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-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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.