[Asterisk-Users] Sip behind the NAT

[Jeffery Chen]

If your Astersik server behind NAT too, your need modify SIP.conf like
this....

externalIP= x.x.x.x
localnet= x.x.x.

hope this can help you....


On 12/8/05, Moises Silva <moises.silva at gmail.com> wrote:
>
> 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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--
Jeffery
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