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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|- [EMAIL PROTECTED]
|ederico Giannici http://www.neomedia.it
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