[Asterisk-Users] SIP conections, with RTP not going trough Asterisk

[Tiago Stein D`Agostini]

So, is there any other option that prevents that from happening? 
Something that I might have turned on  and makes Dial work  trough 
asterisk? I already even removed asterisk completelyu from system and 
reinstalled it to be fresh new... still all RTP goes trough Asterisk 
machine. And the server really can't handle many connections this way.

Thanks for the help.

Peter Bowyer wrote:

>On 17/04/06, Tiago Stein D`Agostini <tiago at khomp.com.br> wrote:
>  
>
>>Hi, sorry to bother again. But I still cannot make it work. I made all
>>acounts have canreinvite=yes, but found no option in Dial aplication to
>>make the phones exchange RTP directly between them.  Can anyone tell me
>>wich option should I look at? I am stuck with this (probably simple)
>>problem for almost a whole week.
>>    
>>
>
>You're trying too hard - unless you tell it not to, the Dial
>application will do what you're asking. As Olle said, this is the
>default.
>
>Peter
>
>--
>Peter Bowyer
>Email: peter at bowyer.org
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