[Freeswitch-users] Remote Phone Routing Update - Now no RTP

[Shawn Lewis]

Ok, I posted previously asking how to figure out the registered endpoint 
of extensions registering.  I used the doublenat.xml sip profile 
solution, then applied the following the dialplan:


     <extension name="public_extensions">
        <condition field="destination_number" expression="1010">
        <action application="set" data="dialed_ext=${destination_number}"/>
          <action application="set" 
data="contact=${sofia_contact(doublenat/${dialed_ext}@X.X.X.X)}"/>
          <action application="bridge" data="sofia/doublenat/${contact}"/>
        </condition>
     </extension>

where X.X.X.X is the EXTERNIP, calls now ring the outside extension, but 
get no voice/rtp.

I have enable the proxy-media setting in my double-nat profile as well.

What happens is the internal extension dials external NAT'd extention 2001.

sofia passes my extrernal IP and PORT successfully in the SDP of the 
Invite to the remote extension.

I receive my 180-ringing.
I receive my 200 OK when answered, in the SDP of the message it is 
providing the 192.168.1.5 address of the IP telephone.

So now FS starts to send media from my IP to 192.168.1.5........  of 
course which will never reach endpoint registered phone...

Here is my doublenat.xml in my sip_profiles, am i missing something?? I 
would think FS would know not to send there.....

<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="doublenat">
  <!-- This profile is only for outbound registrations to providers -->

  <domains>
    <domain name="doublenat" parse="true"/>
  </domains>

  <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="public"/>
    <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="force-register-domain" value=${domain}/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <param name="inbound-late-negotiation" value="true"/>
<!-- <param name="apply-nat-acl" value="rfc1918"/> -->
    <param name="NDLB-force-rport" value="true"/>
    <param name="inbound-proxy-media" value="true"/>
    <!--<param name="enable-3pcc" value="true"/>-->
  </settings>
</profile>
~

Any pointers would be greatly appreciated of course..

Thanks
Shawn



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