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

[Andrei Sotirov]

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