[Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk

[Zen Kato]

Hi,

Thank you for the information. There are "t"s in Dial command in 
extensions.conf. When I deleted these "t"s, each sip phones were
directly communicating. I just wrote these "t"s from the examples.

Does these "t" and "T" are used for transfer(blind/consaltation) from
called user and calling user, respectively? If we don't have these
't' and 'T', can't we do transfer?

Regards,

Zen

"Girish Gopinath" <gopinath_girish at hotmail.com> wrote  :

> Zen,
> 
> >I am trying to confirm the command 'canreinvite=yes' in sip.conf
> >using grandstream BT101/2s and snom100s. In either case, no description
> >nor 'canreinvite=yes', media stream always go through *.
> >
> >Do I need another settings for confirming sip clients directly
> >communicate each other?
> 
> Do you have a Dial statement that has "t" or "T" in it?
> This will force the media stream to pass through Asterisk.
> 
> Regards, Girish
> 
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