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

[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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/cf50ac94/attachment.htm