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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hello Wassim,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> </span><span lang=EN-US style='font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'>server side you can check the
RTP ports configured in rtp.conf which you will find in /etc/asterisk/. If the
file isn’t there, here are the defaults:<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal style='margin-left:35.4pt'><span lang=EN-US
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>;[general]<o:p></o:p></span></p>
<p class=MsoNormal style='margin-left:35.4pt'><span lang=EN-US
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>;<o:p></o:p></span></p>
<p class=MsoNormal style='margin-left:35.4pt'><span lang=EN-US
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>; RTP
start and RTP end configure start and end addresses<o:p></o:p></span></p>
<p class=MsoNormal style='margin-left:35.4pt'><span lang=EN-US
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>;<o:p></o:p></span></p>
<p class=MsoNormal style='margin-left:35.4pt'><span lang=EN-US
style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>;
Defaults are rtpstart=5000 and rtpend=31000<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>You can even debug the RTP : CLI> rtp debug ip
xxx.xxx.xxx.xxx(linksys)<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Asterisk listens on one of those ports(rtp.conf ones) when a
call is initiated. The same does your Linksys GW: it will listen only on the RTP
configured ports.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Check the firewall between the VoIP server and the Linsys GW and
check the firewall on the Asterisk server. <o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Debugging SIP you can see which ports are involved. <o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>There might be other problems, maybe because you are trying to
directly pass the call from one peer(let’s say an external voice provider)
to the other(linksys). In that case careinvite=no is be your friend.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Regards,<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Alex<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Segoe UI","sans-serif"'>Da:</span></b><span
style='font-size:10.0pt;font-family:"Segoe UI","sans-serif"'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>Per conto di </b>wassim
darwich<br>
<b>Inviato:</b> giovedě 28 gennaio 2010 21:41<br>
<b>A:</b> asterisk-users@lists.digium.com<br>
<b>Oggetto:</b> [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short<o:p></o:p></span></p>
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<p class=MsoNormal>Hi:<o:p></o:p></p>
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<p class=MsoNormal>Firewall is disabled ,so no need to worry about
firewall,but i dont know where to check rtp settings and what do i
need to search for ,can you guide me please.<o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif"'><o:p> </o:p></span></p>
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