<blockquote style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;" class="gmail_quote">I&#39;ll always be on the internal network so should never need the nat.<br></blockquote><div>

<br>Try starting FS with the -nonat option. It&#39;ll start faster and skip all the stuff where it tries to find your external IP and open routes through the NAT firewall.<br><br>On your sip profiles set sip-ip and rtp-ip to your internal IP.<br>

<br>Once you&#39;ve done that you should have 2-way audio to anywhere on your LAN.<br><br>-Steve<br><br><br></div><br><br><div class="gmail_quote">On 25 July 2011 05:08, Mitch Johnson <span dir="ltr">&lt;<a href="mailto:mitch.johnson7@gmail.com">mitch.johnson7@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div style="word-wrap:break-word">Sorry for the delayed response, got tied up in other things.  A quick recap. I can dial from my call manager to Freeswitch but cannot dial from freeswitch to call manager.<div>

<br></div><div>I wasn&#39;t sure what you meant by setting the ext-sip-ip and ext-rtp-ip.  So I replaced the auto-nat with local-network-acl.  I&#39;ll always be on the internal network so should never need the nat.</div>

<div><br></div><div>I saw in the acl.conf.xml that it automatically populates all my local subnets.  The networks are correct under the domains of the acl.conf.</div><div><br></div><div>I&#39;m not sure where else to look.  Any help would be greatly appreciated.</div>

<div><br></div><div>Thanks so much,</div><div><br></div><div>Mitch</div><div><br><div><div>On Jul 19, 2011, at 11:29 PM, <a href="mailto:freeswitch-users-request@lists.freeswitch.org" target="_blank">freeswitch-users-request@lists.freeswitch.org</a> wrote:</div>

<br><blockquote type="cite"><span style="border-collapse:separate;font-family:Helvetica;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:-webkit-auto;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;font-size:medium"><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px">

<span style="font-family:Helvetica;font-size:medium;color:rgb(127, 127, 127)"><b>From:<span> </span></b></span><span style="font-family:Helvetica;font-size:medium">Brian West &lt;<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>&gt;<br>

</span></div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px"><span style="font-family:Helvetica;font-size:medium;color:rgb(127, 127, 127)"><b>Subject:<span> </span></b></span><span style="font-family:Helvetica;font-size:medium"><b>Re: [Freeswitch-users] One way calling issue over an internal trunk</b><br>

</span></div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px"><span style="font-family:Helvetica;font-size:medium;color:rgb(127, 127, 127)"><b>Date:<span> </span></b></span><span style="font-family:Helvetica;font-size:medium">July 19, 2011 8:55:02 PM EDT<br>

</span></div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px"><span style="font-family:Helvetica;font-size:medium;color:rgb(127, 127, 127)"><b>To:<span> </span></b></span><span style="font-family:Helvetica;font-size:medium">FreeSWITCH Users Help &lt;<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>&gt;<br>

</span></div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px"><span style="font-family:Helvetica;font-size:medium;color:rgb(127, 127, 127)"><b>Reply-To:<span> </span></b></span><span style="font-family:Helvetica;font-size:medium">FreeSWITCH Users Help &lt;<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>&gt;<br>

</span></div><div><div></div><div class="h5"><br><br>you need to set the ext-sip-ip and ext-rtp-ip and deal with setting local-network-acl to the proper network/mask because you&#39;re answering with RFC1918 media IP&#39;s to the cisco… its retarded and will just start sending to those IP&#39;s and they&#39;ll usually never make it back to you.<div>

<br></div><div>/b</div><div><br><div><div>On Jul 19, 2011, at 7:16 PM, Mitch Johnson wrote:</div><br><blockquote type="cite"><span style="border-collapse:separate;font-family:Helvetica;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:-webkit-auto;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;font-size:medium">I have a connection between my FreeSWITCH and a Cisco CallManager:<br>

<br>This is from the default.xml<br><br>&lt;extension name=&quot;Dial to CM8&quot;&gt;<br>     &lt;condition field=&quot;destination_number&quot; expression=&quot;^(800\d)$&quot;&gt;<br>       &lt;action application=&quot;bridge&quot; data=&quot;<a href="mailto:sofia/internal/$1@172.16.200.100" target="_blank">sofia/internal/$1@172.16.200.100</a>&quot;/&gt;<br>

     &lt;/condition&gt;<br>    &lt;/extension&gt;<br></span></blockquote></div></div></div></div></span><br></blockquote></div><br></div></div><br>_______________________________________________<br>
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