[asterisk-biz] RTP redirect

[Mark]

Correct me if I am wrong, however what he is looking for is an external 
bridge as opposed to a Native bridge???


support wrote:

>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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