[Asterisk-Users] SER & Asterisk with DID incoming and out going

[ram]

Hi

thanks for the reply

ya the default is NAT=YES only

if i keep reinvite=no, the my server b/w consuming lot
since i have bottleneck of server bandwidth

ram


On 3/16/06, Andrei Sotirov <andrei at unixsol.org> wrote:
>
> ram wrote:
> > Hi all
> >
> > I have badly NATed Clients proble with one way Voice
> >
> > After reading some documents people ask me to use STUN Server
> > But still i have some problem with one way Voice
> use stun on dinamic ip :)
> >
> > I have setup like below
> >
> > iam trying with 2 extensions
> >
> > 1 extention in the same LAN where the  * installed
> > 2 extension in different network, NATed IP ,
> > 3. both the side iam use SIPURA
> > 4. i have 2 DID from provider
> > 5. i have route them to appropriate extensions
> >
> > Iam able to make calls in and out
> >
> > but the problem where iam setting up server have limited bandwidth
> > So i have installed G729 codec
> >
> > So i want to make RTP
> >
> > so i made setup caninvite=yes
> >
> canreinvite=no
> nat=yes
> > since my provider support that option
> >
> > then my NAT Clients have One way Voice problem
> >
> > So after Reading some DOCS SER, should be able to do this Job
> >
> > so SER can be integrated with *, if yes
> > can any one point me to some URL
> >
> > thanks
> >
> > ram
> >
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> --
> Поздрави,
> Андрей Сотиров
>
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