asterisk users - Aug 2005 - howto let the stream not passing asterisk

We need to configure asterisk to authenticate two sip ATAs, but the stream must
go directly from one to another ata without tuching asterisk.

Is this possible adding canreinvite=yes into sip.conf?

is it true laso if asterisk doesn't recognize the spd (t38)?

thanks

Rosario
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Rosario Pingaro wrote:
> We need to configure asterisk to authenticate two sip ATAs, but the 
> stream must go directly from one to another ata without tuching asterisk.
>  
> Is this possible adding canreinvite=yes into sip.conf?
>  
> is it true laso if asterisk doesn't recognize the spd (t38)?
>  
> thanks
>  
> Rosario
>  
>
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Hello,
Yes, If they support the same codec and don't put "t" /
"T" with Dial
command on d extensions.conf.
ATA186 has a problem with "canreinvite=yes"
for more info 
http://lists.digium.com/pipermail/asterisk-doc/2004-June/000547.html


Cheers,
~Madhawa