[Asterisk-Users] Sip behind the NAT

[chawki hammoud]

Hi:
i added these two lines to my general context ,but
nothing happened the same result the sound came in one
way for 3 seconds and stopped but it didnt hangup.

--- Jeffery Chen <jeffery9 at gmail.com> wrote:

> 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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> > >
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> 
> --
> Jeffery
> > _______________________________________________
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