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Hi Brian, <br>
Did you have any luck with the EC2 instances. I tried a few things,
but I just cant get RTP to run through it. PLease let me know.<br>
<br>
Best<br>
Damon<br>
<br>
Brian West wrote:
<blockquote
cite="mid:F378D7F6-F3FD-4FB8-BE61-E879E71AFB9A@freeswitch.org"
type="cite">
<pre wrap="">Let me launch mine in 32bit and see what I can do over the weekend...
I had this working without a problem. ;/
/b
On Sep 6, 2008, at 12:09 AM, Damon Brown wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Ive Tried the following with no success:
internal.xml
<param name="ext-rtp-ip" value="75.101.142.208"/>
<param name="ext-sip-ip" value="75.101.142.208"/>
Ive also tried just changing the vars.xml file. I also tried
forwarding the incoming rtp connections to my test pc. all with no
audio success. I am sure im doing something wrong I jsut cant find
it.
I found another suggestion on the wiki of creating a "double nat"
that listens on 5090. That didnt change anything either, here is
that information:
<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5090"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="manage-presence" value="false"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<!--<param name="enable-3pcc" value="true"/>-->
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" value="false"/>
<!-- additional bind parameters for TLS -->
<param name="tls-bind-params" value="transport=tls"/>
<!-- Port to listen on for TLS requests. (5061 will be used if
unspecified)$
<param name="tls-sip-port" value="5081"/>
<!-- Location of the agent.pem and cafile.pem ssl certificates
(needed for $
<param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/>
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may
not work w$
<param name="tls-version" value="tlsv1"/>
</settings>
Maybe someone can see what is going on here.
Thanks,
Damon
-----Original Message-----
From: "Brian West" <a class="moz-txt-link-rfc2396E" href="mailto:brian@freeswitch.org"><brian@freeswitch.org></a>
Sent: Friday, September 5, 2008 6:57pm
To: <a class="moz-txt-link-abbreviated" href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>
Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml
profile on ec2 duplicate them from the external profile.
/b
On Sep 5, 2008, at 8:47 PM, Damon Brown wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Yes, I have all of the valid posts open on my security group
-d
</pre>
</blockquote>
<pre wrap="">Brian West
<a class="moz-txt-link-freetext" href="sip:brian@freeswitch.org">sip:brian@freeswitch.org</a>
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</pre>
</blockquote>
<pre wrap=""><!---->
Brian West
<a class="moz-txt-link-freetext" href="sip:brian@freeswitch.org">sip:brian@freeswitch.org</a>
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