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

2006-03-16 Thread 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 <[EMAIL PROTECTED]> 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 Voiceuse 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=nonat=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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Re: [Asterisk-Users] SER & Asterisk with DID incoming and out going

2006-03-16 Thread 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

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

2006-03-16 Thread ram
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
 
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
 
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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