[Asterisk-Users] Sip behind the NAT

[Bharath]

Forward UDP Ports 10000-20000 to your asterisk box.


On 12/8/05, Jeffery Chen <jeffery9 at gmail.com> wrote:
>
> can u paste your sip.conf general section,,?
>
> there have another possible cause... the both side use different codecm
> and asterisk can not translaste it ...
>
>
> --
> Jeffery
>
>
> On 12/8/05, chawki hammoud <cyhammoud at yahoo.com> wrote:
> >
> > 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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> > > --
> > > Jeffery
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