<html>
<head>
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 12pt;
font-family:Calibri
}
--></style></head>
<body class='hmmessage'><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></body>
</html>