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

[Tiago Stein D`Agostini]

Thanks, that was the problem,  I had the t option on the Dial 
application. Nor that I removed them  it works.

Thank you.

Rich Adamson wrote:

> Tiago Stein D`Agostini 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.
>
>
> The canreinvite=yes is required, however your Dial statements used to 
> complete calls between the sip devices cannot use several of the 
> options including t, T, etc.
>
> If you remove all options from the Dial statement, restart asterisk, 
> and place a test call, those sip phones that can "see" each other will 
> auto-negotiate rtp directly between them.
>
> If they cannot see each other (eg, nat or firewalls involved), they 
> will not auto-negotiate direct rtp.
>
> There is no option for you to specify to "forced" direct rtp.
>
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