<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><div>Hi Todd,</div><div>From my experience one-way or no-way audio is typically caused by the Asterisk system being behind a NAT or improperly configured firewall (usually NAT). </div><div><br></div><div>The provider you are dealing with is correct when they say it doesn’t matter if they listen on different RTP ports. That all gets negotiated at the beginning of the call. </div><div><br></div><div>Chad</div><br><div><div>On Jul 2, 2014, at 1:49 PM, Todd R. <<a href="mailto:tjrlist@live.com">tjrlist@live.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div class="hmmessage" style="font-size: 12pt; font-family: Calibri; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;"><div dir="ltr">Been working with Asterisk for a long time but this is the first time I have dealt with this issue.<div><br></div><div>I am setting up an Asterisk box (FreePBX not my choice) to interface with an e911 provider.</div><div><br></div><div>They say their switches only listen for RTP on ports 20000-21001 which is outside the normal range Asterisk listens on 10000-20000.</div><div><br></div><div>I wish I knew more about this topic but since I have never had an issue interfacing with providers, ITSP etc., I just haven't had a need to know.</div><div><br></div><div>I get audio on some calls and others not so much.</div><div><br></div><div>How do I deal with this?</div><div><br></div><div>I don't really want to change the RTP ports that Asterisk listens on because this is a production system with trunks pointing to several other providers etc.</div><div><br></div><div>The 911 provider says I don't need to listen on 20000-21001, that's just what they listen on.</div><div><br></div><div>In fact, they say this exactly "<span style="font-size: 12pt;">You can listen on what you want, as long as the RTP port is sent to us in the INVITE SDP info.".</span></div><div><span style="font-size: 12pt;"><br></span></div><div><span style="font-size: 12pt;">Any assistance with solving this issue would be greatly appreciated, I have done my digging in Google etc before asking here as always.</span></div><div><span style="font-size: 12pt;"><br></span></div><div><span style="font-size: 12pt;">Thanks in advance.</span></div></div>--<span class="Apple-converted-space"> </span><br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by<span class="Apple-converted-space"> </span><a href="http://www.api-digital.com/">http://www.api-digital.com</a><span class="Apple-converted-space"> </span>--<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></div></blockquote></div><br></body></html>