Hi
Although canreinvite option is yes, the asterix doesn't send reinvite and
the media is going through the asterix instead of between the two sip phones.
Both sip phones (handytone 486) don't use NAT and are configure with
canreinvite option yes and use the same codec G.729. And Dial() command
don't contains t or T.
Any suggestion on what could be the problem ?
Thanks,
Ishay
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