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Level 3 uses port 5070 for their sip server but requires port 5060 to be used on our side; I made changes to the vars.xml config file to support this. I was just able to get the server to answer a call by changing the register value to false and adding an entry
 in the ACL config for the Level 3 server however now it seems that the audio isn't working correctly. I'm using a simple dial plan to echo the audio back but all I get is dead air.
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&nbsp; &nbsp;&nbsp;&lt;<span style="color: #b22221">extension</span> <span style="color: #ff1b18">name</span>=<span style="color: #000000">&quot;</span>level3<span style="color: #000000">&quot;</span>&gt;</div>
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&nbsp; &nbsp; &nbsp; &lt;<span style="color: #b22221">condition</span> <span style="color: #ff1b18">
field</span>=<span style="color: #000000">&quot;</span>destination_number<span style="color: #000000">&quot;</span>
<span style="color: #ff1b18">expression</span>=<span style="color: #000000">&quot;</span>^13127567000$<span style="color: #000000">&quot;</span>&gt;</div>
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&nbsp; &nbsp; &nbsp; &nbsp; &lt;<span style="color: #b22221">action</span> <span style="color: #ff1b18">
application</span>=<span style="color: #000000">&quot;</span>answer<span style="color: #000000">&quot;</span>/&gt;</div>
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&nbsp; &nbsp; &nbsp; &nbsp; &lt;<span style="color: #b22221">action</span> <span style="color: #ff1b18">
application</span>=<span style="color: #000000">&quot;</span>echo<span style="color: #000000">&quot;</span> /&gt;</div>
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<span style="color: #2c26f9">&nbsp; &nbsp; &nbsp; &lt;/</span>condition<span style="color: #2c26f9">&gt;</span></div>
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<span style="color: #2c26f9">&nbsp; &nbsp; &lt;/</span>extension<span style="color: #2c26f9">&gt;</span></div>
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<div>On Sep 11, 2011, at 7:47 PM, Nandy Dagondon wrote:</div>
<br class="Apple-interchange-newline">
<blockquote type="cite">in addition to peter's advise, take a look at the SIP port 5070. FS is using port 5080 for the external SIP profile. modify the port number at &quot;external.xml&quot; then delete the port numbers in your proxy settings.<br>
<br>
<div class="gmail_quote">On Sun, Sep 11, 2011 at 6:36 PM, Peter Olsson <span dir="ltr">
&lt;<a href="mailto:peter.olsson@visionutveckling.se">peter.olsson@visionutveckling.se</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
You're not giving us much information here. Please post exactly what doesn't work, and also pastebin the actual logs from FreeSWITCH.<br>
<br>
/Peter<br>
________________________________________<br>
Från: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] f&amp;#246;r Chad Vogel [<a href="mailto:cvogel@lyonl.com">cvogel@lyonl.com</a>]<br>
Skickat: den 11 september 2011 03:28<br>
Till: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Ämne: [Freeswitch-users] external sip profile<br>
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hello,<br>
<br>
I'm trying to switch from asterisk to freeswitch; however i'm wondering how can I create a sip profile because the sip profile i created doesn't seem to function with level 3.<br>
<br>
here is the sip profile i created that isn't working:<br>
<br>
&lt;include&gt;<br>
&nbsp;&lt;gateway name=&quot;Level3&quot;&gt;<br>
&nbsp; &nbsp;&lt;param name=&quot;username&quot; value=&quot;USERNAME&quot;/&gt;<br>
&nbsp; &nbsp;&lt;param name=&quot;password&quot; value=&quot;PASSWORD&quot;/&gt;<br>
&nbsp; &nbsp;&lt;param name=&quot;proxy&quot; value=&quot;<a href="http://4.55.35.60:5070/" target="_blank">4.55.35.60:5070</a>&quot;/&gt;<br>
&nbsp; &nbsp;&lt;param name=&quot;expire-seconds&quot; value=&quot;3600&quot;/&gt;<br>
&nbsp; &nbsp;&lt;param name=&quot;register-transport&quot; value=&quot;udp&quot;/&gt;<br>
&nbsp; &nbsp;&lt;param name=&quot;register&quot; value=&quot;true&quot;/&gt;<br>
&nbsp;&lt;/gateway&gt;<br>
&lt;/include&gt;<br>
<br>
here is my asterisk profile (it works):<br>
<br>
[level3_out]<br>
type=peer<br>
nat=no<br>
host=4.55.35.60<br>
username=***Username***<br>
secret=***Password***<br>
dtmfmode=rfc2833<br>
port=5070<br>
<br>
[level3_in]<br>
nat=no<br>
insecure=very<br>
dtmfmode=rfc2833<br>
disallow=all<br>
context=from-trunk<br>
canreinvite=no<br>
allow=ulaw&amp;alaw<br>
host=4.55.35.60<br>
type=peer<br>
port=5070<br>
<br>
How can I create a sip profile that will function the same in freeswitch?<br>
<br>
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!DSPAM:4e6c82af32761635315745!<br>
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</blockquote>
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