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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Right after I hit send I saw that in SRC there was a 2<sup>nd</sup> rtp.conf example that listed what you have sent. I have added this to my RTP.conf file and
and added these ports to my router config. I’ll be out of the house and will check later today. Thanks for your help.<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>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:#17365D">Bill Hurlock<o:p></o:p></span></b></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> Robert A. Poff WB3AWJ [mailto:wb3awj@comcast.net]
<br>
<b>Sent:</b> Wednesday, February 12, 2014 11:10 AM<br>
<b>To:</b> Bill Hurlock<br>
<b>Cc:</b> app rpt-users<br>
<b>Subject:</b> Re: [App_rpt-users] SIP Help Needed<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><span style="font-family:"Arial","sans-serif";color:black">Here's what is in mine verbatim.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Arial","sans-serif";color:black">Ah....... I just looked into an ACID install of mine. Didn't see rtp.conf there at all. You may need to create it.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Arial","sans-serif";color:black">When I first starting to add extensions, I used the Asterisk book as a guide. Very helpful.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Arial","sans-serif";color:black">Here's the online version :
<a href="http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.html">
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/index.html</a><o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Arial","sans-serif";color:black">It covers Asterisk 1.8, but most of the config stuff still applies to 1.4 I find.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Arial","sans-serif";color:black"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Arial","sans-serif";color:black">Anyhow, here's what I have in rtp.conf on my node that has SIP phones attached:<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><span style="font-family:"Arial","sans-serif";color:black">;<br>
; RTP Configuration<br>
;<br>
[general]<br>
;<br>
; RTP start and RTP end configure start and end addresses<br>
; These are the addresses where your system will RECEIVE audio and video stream$<br>
; If you have connections across a firewall, make sure that these are open.<br>
;<br>
<br>
rtpstart=10000<br>
rtpend=10999<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Arial","sans-serif";color:black"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Arial","sans-serif";color:black">And here's a link to a page concerning rtp.conf :
<a href="http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf">http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf</a><o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Arial","sans-serif";color:black"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="font-family:"Arial","sans-serif";color:black">Hope this helps.<o:p></o:p></span></p>
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