[Asterisk-Users] Sip behind the NAT

[Moises Silva]

what type of NAT do you have? sync? full cone? cone restricted, port
restricted?
any messages in asterisk verbose console?

best regards

On 12/7/05, chawki hammoud <cyhammoud at yahoo.com> wrote:
>
> Hi list:
> i have an asterisk box behind the NAT ,when i try to
> send calls through Sip to the voip provider server the
> call is answered but in a one way calling,I hear  the
> voice of the other side just for 4 seconds and then
> stop but the call do not hangup.
>
> my sip.conf is:
> [voip provider]
> type=peer
> host=213.112.50.8
> username=XXXXXXX
> secret=XXXXXX
> fromuser=XXXXXXX
> canreinvite=no
> nat=yes
> insercure=invite
> disallow=all
> allow=gsm
>
>
>
>
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