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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Sorry to bug you guys.&nbsp; I figured it
out.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>In case anyone else is just learning to
crawl with freeswitch&#8230;.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I enabled the following in the
sip_profiles to get around the authorization errors (for now):<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp; &lt;!--&nbsp; comment
the next line and uncomment one or both of the other 2 lines for call
authentication --&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp; &lt;param
name=&quot;accept-blind-reg&quot; value=&quot;true&quot;/&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp; &lt;!-- accept any
authentication without actually checking (not a good feature for most people)
--&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp; &lt;param
name=&quot;accept-blind-auth&quot; value=&quot;true&quot;/&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Then I started receiving a 404 route not
found so I modified the public dialplan with the following:<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp; &lt;extension
name=&quot;public_call&quot;&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&lt;condition field=&quot;destination_number&quot;
expression=&quot;^(.*)$&quot;&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&lt;action application=&quot;bridge&quot;
data=&quot;sofia/gateway/gafachi/$1&quot;/&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;/condition&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp; &lt;/extension&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Then I wasn&#8217;t getting 2-way audio so
I changed the sip profile for nat (which I&#8217;m using internally) and set
the ext-sip-ip and the ext-rtp-ip to the same value as the rtp-ip and the
sip-ip (since I&#8217;m only using for internal nat through firewall to sip
provider):<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&lt;!--&nbsp;&nbsp;&nbsp; &lt;param
name=&quot;ext-rtp-ip&quot; value=&quot;$${external_rtp_ip}&quot;/&gt; --&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&lt;!--&nbsp;&nbsp;&nbsp; &lt;param
name=&quot;ext-sip-ip&quot; value=&quot;$${external_sip_ip}&quot;/&gt; --&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp; &lt;param
name=&quot;ext-rtp-ip&quot; value=&quot;$${local_ip_v4}&quot;/&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp; &lt;param
name=&quot;ext-sip-ip&quot; value=&quot;$${local_ip_v4}&quot;/&gt;<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>And now I have calls routed by sipx to
freeswitch and through the firewall to our internet sip provider.&nbsp;
Obviously the current configuration isn&#8217;t secure but it&#8217;s enough to
get things going. <o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] <b><span
style='font-weight:bold'>On Behalf Of </span></b>Jay Reeder<br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, April 24, 2008
4:40 PM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">freeswitch-users@lists.freeswitch.org</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> [Freeswitch-users]
Problems with initial setup - basic nat</span></font><o:p></o:p></p>

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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>We&#8217;re setting up a SipXecs server in-house to manage
about 20-30 polycom sip phones.&nbsp; We have an Audiocodes Mediant 2000 to use
as a gateway but for testing I was also trying to setup sip in/out dialing
through the firewall.&nbsp; I&#8217;ve wanted a reason to start playing with
freeswitch so I thought this would be an excellent opportunity to use
freeswitch for the Nat traversal.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I&#8217;ve been through the wiki and reviewed list archives
but I&#8217;m missing something.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I have RC3 on Centos (initially a trixswitch load but then
upgraded to the new RC3) with the standard config files.&nbsp; I did remove the
older ones and re-installed the samples.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>This is a pretty basic install with a gafachi gateway setup
for the outbound sip profile, and the firewall&#8217;s external ip setup for
the external_rtp and external_sip values (in vars.xml), and the firewall port
forwards all recommended ports(from wiki getting started page) into freeswitch.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>This is where I&#8217;m stuck.&nbsp; I have sipx attempting
to send calls to Freeswitch on port 5070 (for nat) but Freeswitch won&#8217;t
accept the call and is logging: <o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>2008-04-24 16:20:26 [DEBUG] sofia.c:219
sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication
Required] session: n/a<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>The nat sip_profile is setup per default to answer port 5070
and authentication (per default) is disabled.&nbsp; <o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I&#8217;m sure it&#8217;s something obvious but what am I
missing?<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Thanks,<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Jay<o:p></o:p></span></font></p>

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