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
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