[asterisk-users] RTP traffic through Asterisk??

Leonja Cerebro liosf7 at gmail.com
Mon Nov 16 00:52:13 CST 2009


see the DTMF method on both phones.

2009/11/14 Ignacio <sanfermines at gmail.com>

> Ok, thank you very much. I will try to find any information in
> asterisk documentation about RTP.
>
> On Fri, Nov 13, 2009 at 3:03 PM, John A. Sullivan III
> <jsullivan at opensourcedevel.com> wrote:
> > 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
> > jsullivan at opensourcedevel.com
> >
> > http://www.spiritualoutreach.com
> > Making Christianity intelligible to secular society
> >
> >
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