[Users] inbound call failed

[unplug]

I think that it is a NAT problem.  Found from some document showing
that using STUN is not stable much.  As I know, openser will have a
nathelper module to process the NAT.  There is an configuration
example in the source "nathelper.cfg".  Does it work to solve the NAT
problem instead of using STUN?

On 12/13/05, Klaus Darilion <klaus.mailinglists at pernau.at> wrote:
> unplug wrote:
> >>From the log, it showed lookup(): 'account name' Not found in usrloc.
> > I think it is the NAT problem, so I use stun for the case below.
>
> Does the INVITE from the GW does have a proper formated request URI? The
>   called phone number must be mapped to the username before doing
> lookup("location"). E.g. you can use aliases to map phone numbers to
> user names lookup("aliases"); (or use the aliasdb module)
>
> regards
> klaus
>
>
> >
> > Telephone (A)
> >   |
> > PSTN
> >   |
> > G/W
> >   |
> > openser
> >   |
> > NAT -- IP phone(B) -- STUN
> >   + ----- IP phone(C) ------+
> >
> > Below is the result:
> > A to B/C is ok
> > B/C to A is ok
> > However, there is no sound when B to C or vice versa.  What reason
> > will cause no sound between B and C?  Is the the reason from the
> > NAT/STUN?
> >
> >
> > On 12/12/05, Klaus Darilion <klaus.mailinglists at pernau.at> wrote:
> >
> >>Use ngrep to watch for incoming SIP requests on the SIP proxy.
> >>
> >>Take a look at the logfiles on the gateway.
> >>
> >>klaus
> >>
> >>unplug wrote:
> >>
> >>>Below is the common configuration of the network.
> >>>
> >>>Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger)
> >>>
> >>>I can make call from softphone to Telephone.  However, it is failed to
> >>>make call from Telephone to softphone.  I wonder why it happened and
> >>>any reference to trace the problem.  Anyone have such experience?
> >>>
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> >>>
> >>>
> >>
> >>
> >
> >
>
>