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<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> </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 ‘remove’ 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 ‘tons’ of users look at
something like SER. 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> </o:p></span></font></p>
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<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> </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’s
are passing through my asterisk. How could I achieve that they go directly from
phone to phone? 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> </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> </o:p></span></font></p>
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