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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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 < [EMAIL PROTECTED]>
> 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 <[EMAIL PROTECTED]>
> 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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