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) 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050828/9b4047d4/attachment.htm
