[Asterisk-Users] Sip behind the NAT

[Jeffery Chen]

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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