[asterisk-biz] RTP redirect

support brian at ezzitel.com
Thu Feb 16 07:51:20 MST 2006


Let me rephrase my previous answer..  canreinvite is the answer HOWEVER
if there is a codec negotiation problem between the endpoints asterisk
will take over and transcode the media.  My only suggestion to you is to
watch the debug of the invites and see what its doing.


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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