asterisk users - Nov 2005 - Re-invite don't always work

I want to be SURE that two UAs connected by asterisk (1.2-beta2) use a 
direct RTP stream, so that they don't waste the bandwidth of asterisk.

How can I obtain it?

I have set "canreinvite=yes", but I have read that in this case
asterisk
TRY to do a reinvite, but if it don't succeed, it remains "in the 
middle". Is it right?

Looking at the output of a tcpdump it seems that actually it doesn't 
work in any condition.

We have a Cisco PSTN gateway that calls the asterisk, witch forward the 
call to one of two phones.

In the case of an analog phone attached to a "Fritz! Box Fon WLAN", it
seems that the RTP stream don't flow through asterisk.

In the case of a Grandstream GXP-2000, it seams that it sends its RTP 
stream directly to the gateway BUT the gateway keeps sending its RTP 
stream through asterisk!

Anybody knows why it happens?

How can I avoid this?

How can i FORCE asterisk to ALWAYS reinvite the calls?
I prefer the call to NOT be established instead of flowing through asterisk.


Thanks.

-- 
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    |-                      giannici@neomedia.it
    |ederico Giannici      http://www.neomedia.it
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Federico Giannici wrote:
> How can i FORCE asterisk to ALWAYS reinvite the calls?
> I prefer the call to NOT be established instead of flowing through 
> asterisk.
You can't.