On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote: > I have just established a call between 2 sip phones and I have noticed > that all RTP traffic goes through Asterisk Server. > > I was expecting RTP traffic went to one phone to another phone directly. > > I set canreinvite=yes in sip.conf in both sip peers. > > I also tested it with 2 mgcp phones and same result, all rtp traffic > goes through Asterisk. > > Is there any way to force traffic to go from one phone to another? <snip> I don't recall where it is off-hand but, somewhere in the Asterisk documentation, there is an explanation of how Asterisk makes a decision about reinvites. You may want to look at that to see if your environment satisfies all the requirements and how it can be adapted if it does not - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com
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