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

[Nick Cameo]

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.
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