<div dir="ltr">see the DTMF method on both phones.<br><br><div class="gmail_quote">2009/11/14 Ignacio <span dir="ltr"><<a href="mailto:sanfermines@gmail.com">sanfermines@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Ok, thank you very much. I will try to find any information in<br>
asterisk documentation about RTP.<br>
<div><div></div><div class="h5"><br>
On Fri, Nov 13, 2009 at 3:03 PM, John A. Sullivan III<br>
<<a href="mailto:jsullivan@opensourcedevel.com">jsullivan@opensourcedevel.com</a>> wrote:<br>
> On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote:<br>
>> I have just established a call between 2 sip phones and I have noticed<br>
>> that all RTP traffic goes through Asterisk Server.<br>
>><br>
>> I was expecting RTP traffic went to one phone to another phone directly.<br>
>><br>
>> I set canreinvite=yes in sip.conf in both sip peers.<br>
>><br>
>> I also tested it with 2 mgcp phones and same result, all rtp traffic<br>
>> goes through Asterisk.<br>
>><br>
>> Is there any way to force traffic to go from one phone to another?<br>
> <snip><br>
> I don't recall where it is off-hand but, somewhere in the Asterisk<br>
> documentation, there is an explanation of how Asterisk makes a decision<br>
> about reinvites. You may want to look at that to see if your<br>
> environment satisfies all the requirements and how it can be adapted if<br>
> it does not - John<br>
> --<br>
> John A. Sullivan III<br>
> Open Source Development Corporation<br>
> +1 207-985-7880<br>
> <a href="mailto:jsullivan@opensourcedevel.com">jsullivan@opensourcedevel.com</a><br>
><br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>We never did too much talking anyway<br>So don't think twice, it's all right<br>----------------------------------------------------------<br>There are more things in heaven and earth, Horatio,<br>
Than are dreamt of in your philosophy.<br>
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