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

[Mojo with Horan & Company, LLC]

correcting an error cause by my own ambiguity:

Mojo with Horan & Company, LLC wrote:
> clarifying that you CANNOT put t or T in there if you want 
> canreinvite=no to have no effect.
you cannot put t or T in there if you want canreinvite=no to have ANY 
effect.  If you want the stream to skip asterisk, and first you've told 
it not to allow reinvites with this canreinvite option, then you have to 
make sure asterisk isn't also being TOLD to listen in on the stream for 
transfer requests (t and T)

Moj

> 
> 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
>>
>>
>>
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-- 
Mojo <mojo at horanappraisals.com>
Office Manager, Horan & Company, LLC
(907) 747-6666 x112