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

[Eric Wieling]

t and T are for "#" transfers.  Other types of transfer are done in
other ways.  Zap FLASH transfers are set in /etc/asterisk/zapata.conf. 
I don't know how you enable/disable SIP or other types of transfers.

On Thu, 2004-03-04 at 06:51, Zen Kato wrote:
> 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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-- 
For Asterisk PBX related documentation go to
http://www.digium.com/index.php?menu=documentation and look at the
"Unofficial Links" section also see
http://www.voip-info.org/wiki-Asterisk also see my site at
http://www.fnords.org/~eric/asterisk/

BTEL Consulting