<?xml version="1.0" encoding="UTF-8"?><rss xmlns:dc="http://purl.org/dc/elements/1.1/" xmlns:content="http://purl.org/rss/1.0/modules/content/" xmlns:atom="http://www.w3.org/2005/Atom" version="2.0"><channel><title><![CDATA[No Audio on Remote VoIP Phones]]></title><description><![CDATA[<p dir="auto">Hi all,</p>
<p dir="auto">I hope so one can help me out here I have Trixbox 2.6.2.3 behind PFsense 1.2.3 My Asterisk box connects to all Trunks and call work fine and i can register a Billion 7404 and a nokia n95 but there is not audio in either direction i.</p>
<p dir="auto">In PF Sense I have AON enable with -Interface =Wan<br />
                                                  Source -Type = Network<br />
                                                  Address -192.168.100.0 &lt;- my internal IP Range<br />
                                                  Source Port - Blank</p>
<p dir="auto">Destination - All fields blank</p>
<p dir="auto">Translation-Address - Interface Address<br />
                                                  port- blank</p>
<p dir="auto">No XMLRPC Snyc not checked</p>
<p dir="auto">Static port enabled</p>
<p dir="auto">I have forwarded all the normal ports 10002-2000  5060 -5082 to trixbox</p>
<p dir="auto">Any help would be great</p>
<p dir="auto">ps it all worked behind IP Cop but a much prefer PF Sense</p>
]]></description><link>https://forum.netgate.com/topic/30403/no-audio-on-remote-voip-phones</link><generator>RSS for Node</generator><lastBuildDate>Mon, 08 Jun 2026 21:05:53 GMT</lastBuildDate><atom:link href="https://forum.netgate.com/topic/30403.rss" rel="self" type="application/rss+xml"/><pubDate>Tue, 08 Feb 2011 01:53:16 GMT</pubDate><ttl>60</ttl><item><title><![CDATA[Reply to No Audio on Remote VoIP Phones on Thu, 10 Feb 2011 02:10:01 GMT]]></title><description><![CDATA[<p dir="auto">I should have mentioned in the newer version of Trixbox has<br />
sip_general_custom.conf which is used for that info as far as i know<br />
I have added it to SIP_NAT.conf did not make any difference</p>
<p dir="auto">cheers</p>
]]></description><link>https://forum.netgate.com/post/265042</link><guid isPermaLink="true">https://forum.netgate.com/post/265042</guid><dc:creator><![CDATA[flaknet]]></dc:creator><pubDate>Thu, 10 Feb 2011 02:10:01 GMT</pubDate></item><item><title><![CDATA[Reply to No Audio on Remote VoIP Phones on Thu, 10 Feb 2011 00:12:51 GMT]]></title><description><![CDATA[<p dir="auto">In sip_nat.conf you need the following lines:</p>
<pre><code>externalip=xxx.xxx.xxx.xxx
localnet=192.168.1.0/255.255.255.0
</code></pre>
<p dir="auto">Replace the xxx.xxx with your public IP address and modify your internal subnet accordingly.</p>
]]></description><link>https://forum.netgate.com/post/265033</link><guid isPermaLink="true">https://forum.netgate.com/post/265033</guid><dc:creator><![CDATA[Gob]]></dc:creator><pubDate>Thu, 10 Feb 2011 00:12:51 GMT</pubDate></item><item><title><![CDATA[Reply to No Audio on Remote VoIP Phones on Wed, 09 Feb 2011 23:10:51 GMT]]></title><description><![CDATA[<p dir="auto">Thanks for help on this</p>
<p dir="auto">Hmmm…<br />
The only other things I would check are the static port on the AON and SIP_Nat.conf.</p>
<p dir="auto">So when you check static port on AON i have static port checked but no port specified is that correct ?</p>
<p dir="auto">SIP_Nat.conf is empty in RTP.conf i have -</p>
<p dir="auto">; RTP Configuration<br />
;<br />
[general]<br />
;<br />
; RTP start and RTP end configure start and end addresses<br />
;<br />
rtpstart=10000<br />
rtpend=20000</p>
<p dir="auto">cheers</p>
]]></description><link>https://forum.netgate.com/post/265026</link><guid isPermaLink="true">https://forum.netgate.com/post/265026</guid><dc:creator><![CDATA[flaknet]]></dc:creator><pubDate>Wed, 09 Feb 2011 23:10:51 GMT</pubDate></item><item><title><![CDATA[Reply to No Audio on Remote VoIP Phones on Wed, 09 Feb 2011 07:22:41 GMT]]></title><description><![CDATA[<p dir="auto">Hmmm…<br />
The only other things I would check are the static port on the AON and SIP_Nat.conf.<br />
If they are correct, you have me stumped.</p>
]]></description><link>https://forum.netgate.com/post/264879</link><guid isPermaLink="true">https://forum.netgate.com/post/264879</guid><dc:creator><![CDATA[Gob]]></dc:creator><pubDate>Wed, 09 Feb 2011 07:22:41 GMT</pubDate></item><item><title><![CDATA[Reply to No Audio on Remote VoIP Phones on Wed, 09 Feb 2011 03:43:31 GMT]]></title><description><![CDATA[<p dir="auto">I have ports 10000 to 20000 open on mine which works fine. You mention 10002 to 2000 - is that a typo?</p>
<p dir="auto">No its not a typo Webmin uses port 10000 i tried to forward 10001 but i think PF Sense doesn't allow ports forwarded to end in an odd number</p>
<p dir="auto">I have set it to 10000 -20000 now as i gather you think this might be causing an issue  still no go</p>
]]></description><link>https://forum.netgate.com/post/264860</link><guid isPermaLink="true">https://forum.netgate.com/post/264860</guid><dc:creator><![CDATA[flaknet]]></dc:creator><pubDate>Wed, 09 Feb 2011 03:43:31 GMT</pubDate></item><item><title><![CDATA[Reply to No Audio on Remote VoIP Phones on Tue, 08 Feb 2011 13:16:15 GMT]]></title><description><![CDATA[<p dir="auto">has your public IP changed since using the IPcop?<br />
Do you have the correct IP info in sip_nat.conf on your trixbox?</p>
<p dir="auto">I have ports 10000 to 20000 open on mine which works fine. You mention 10002 to 2000 - is that a typo?</p>
]]></description><link>https://forum.netgate.com/post/264760</link><guid isPermaLink="true">https://forum.netgate.com/post/264760</guid><dc:creator><![CDATA[Gob]]></dc:creator><pubDate>Tue, 08 Feb 2011 13:16:15 GMT</pubDate></item></channel></rss>