[Asterisk-Users] howto let the stream not passing asterisk

[Rosario Pingaro]

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050808/50a790c0/attachment.htm