[asterisk-users] How to make RTP does not go thru asterisk server

[Mojo with Horan & Company, LLC]

I'm not clear on your usage of t and T in the dial command, so 
clarifying that you CANNOT put t or T in there if you want 
canreinvite=no to have no effect.

Anuj Jain wrote:
> Hi All
> I am using trixbox asterisk 1.2
> I have enabled canreinvite=yes and no  "tT" in the dialplan as it has 
> been described in the various forums.
> Still the voice call goes thru the asterisk server.
> How can i really make the call between 2 grandstream devices( i am using 
> HT 488, HT286 and SIP extensions) after the initial handshake.
> 
> Thanks & Regards
> !DSPAM:500,4524455a101385315134984!
> 
> 
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-- 
Mojo <mojo at horanappraisals.com>
Office Manager, Horan & Company, LLC
(907) 747-6666 x112