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

[Rich Adamson]

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.