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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