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

[Girish Gopinath]

Hi Zen,

>From: Zen Kato <zenkato at pis.bekkoame.ne.jp>
<snip>
>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?

'T' and 't' are used for transfer using #

The 'T' allows the calling user to transfer the call.
't' allows the called user to transfer the call.

Andy Powell's guide to Asterisk http://www.automated.it/guidetoasterisk.htm 
has these details, It is simple, and contains some good basic things about 
Asterisk.

Regards, Girish

>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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