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

[Paul Belanger]

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.
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