<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns="http://www.w3.org/TR/REC-html40">

<head>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<meta name=Generator content="Microsoft Word 11 (filtered medium)">
<!--[if !mso]>
<style>
v\:* {behavior:url(#default#VML);}
o\:* {behavior:url(#default#VML);}
w\:* {behavior:url(#default#VML);}
.shape {behavior:url(#default#VML);}
</style>
<![endif]-->
<style>
<!--
 /* Font Definitions */
 @font-face
        {font-family:Tahoma;
        panose-1:2 11 6 4 3 5 4 4 2 4;}
@font-face
        {font-family:Verdana;
        panose-1:2 11 6 4 3 5 4 4 2 4;}
 /* Style Definitions */
 p.MsoNormal, li.MsoNormal, div.MsoNormal
        {margin:0in;
        margin-bottom:.0001pt;
        font-size:12.0pt;
        font-family:"Times New Roman";}
a:link, span.MsoHyperlink
        {color:blue;
        text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
        {color:purple;
        text-decoration:underline;}
p.MsoAutoSig, li.MsoAutoSig, div.MsoAutoSig
        {margin:0in;
        margin-bottom:.0001pt;
        font-size:12.0pt;
        font-family:"Times New Roman";}
span.EmailStyle18
        {mso-style-type:personal;
        font-family:Arial;
        color:windowtext;}
span.EmailStyle19
        {mso-style-type:personal-reply;
        font-family:Arial;
        color:navy;}
@page Section1
        {size:8.5in 11.0in;
        margin:1.0in 1.25in 1.0in 1.25in;}
div.Section1
        {page:Section1;}
-->
</style>

</head>

<body lang=EN-US link=blue vlink=purple>

<div class=Section1>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Look at canreinvite= in the sip.conf.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>If you &#8216;remove&#8217; Asterisk from
the stream them you are using Asterisk more like a Proxy and less like a PBX.
If this is the case and you want to support &#8216;tons&#8217; of users look at
something like SER.&nbsp; Asterisk is not a Sip proxy but rather a PBX and Media
transcodeing gateway <o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>

<hr size=2 width="100%" align=center tabindex=-1>

</span></font></div>

<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>bijan<br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, December 23, 2004
5:46 PM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Asterisk-Users] rtp
channels not through asterisk</span></font><o:p></o:p></p>

</div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>In wiki pages it is stated that </span></font><font size=1
color=black face=Verdana><span style='font-size:9.0pt;font-family:Verdana;
color:black'>The audio channels (RTP) may go directly from phone to phone or
may go through Asterisk's media bridge.</span></font><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Currently with my settings, I notice that all rtp&#8217;s
are passing through my asterisk. How could I achieve that they go directly from
phone to phone?&nbsp; I assume this way, my machine will have less load and
therefore could handle more calls.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>regards<o:p></o:p></span></font></p>

<p class=MsoAutoSig><b><font size=3 face="Times New Roman"><span
style='font-size:12.0pt;font-weight:bold'>Bijan Karimi</span></font></b><o:p></o:p></p>

<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

</body>

</html>