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

[Tiago Stein D`Agostini]

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.

Thanks for any help.

Ronald Wiplinger wrote:

> Tiago Stein D`Agostini wrote:
>
>> Hi,
>>
>>   Ie been looking for some time how to use asterisk  to initiate SIP 
>> connections between 2 IP phones,  but afetr initiated the 
>> communication making the RTP go directly from one telephone to the 
>> other, without passing by asterisk. Unfortunately I found no 
>> explanations of how to do it.
>>
>> Does anyone care to give a pointer to any explanation about how to do 
>> it?
>>
> canreinvite=yes
> and look at the options for dial()
>
>>
>> Thanks in advance
>>
>
>
> bye
>
> Ronald Wiplinger
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