Nick Cameo
2014-Jan-06 14:27 UTC
[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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140106/3174fc6e/attachment.html>
Eric Wieling
2014-Jan-06 14:38 UTC
[asterisk-users] Dropped call on new CISCO router for no reason!
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.
Paul Belanger
2014-Jan-06 18:33 UTC
[asterisk-users] Dropped call on new CISCO router for no reason!
On 14-01-06 09:27 AM, Nick Cameo wrote:> 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. > > >Show us the problem, give us a SIP trace[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger