[asterisk-biz] RTP redirect

[support]

So you want asterisk not to proxy the media..  Simple.  Put
canreinvite=yes in your peer in sip.conf.  It will do it automatically.



On Fri, 2006-02-17 at 01:19 +1100, Zafer Khodr wrote:
> I am trying to setup asterisk as shown below.
> 
> 
> 				 
>                         xxx.xxx.xxx.xxx
> 		     /=====(Call Origination)====\
> 		    /		      |               \
> 	       R /		      |S               \
> 	      T /			|I                \ R
> 	     P /			|P                 \ T
>        	/			|                   \ P
> 	   S /		   Asterisk                \ 
> 	  T /		            /\                    \ S
> 	 R /			     /  \                    \ T
> 	E /		          /    \                    \ R
>      A /                   /      \                    \ E
>     M /                   /        \                    \ A
>      /                   /          \                    \ M
>     /                   /            \                    \
>    /                   /              \                    \
>   /         __________/                \__________          \
>  /         |					        |          \
> /	     |					        |           \
> yyy.yyy.yyy.yyy					    zzz.zzz.zzz.zzz
> Terminating Gateway 1				 Terminating Gateway 2
> 
> 
> 
> The main objective to to have asterisk in the path of the call but the RTP
> to go directly between the originating and terminating IP's.
> I have had a play around with canreinvite but doesn't seem to make a
> difference.
> 
> If someone could please help me out by pointing me in the right direction,
> that would be great
> 
> Regards,
> Zafer
> 
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